[SR-Users] cisco spa502g issues
Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@ Contact:: sip:5546@:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018@10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp::5060 Methods:: 6815 AOR:: 5...@sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db@10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip::1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp::5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cisco spa502g issues
Hi Daniel, I shall have a look at that when Im back in the office on monday :) Cheers, Ben. On 17/03/17 18:04, Daniel-Constantin Mierla wrote: Hello, maybe the new phones do STUN and or the ALG breaks somehow the signaling. You should send the ngrep output taken on sip server for such a call in order to be able to analyze what can happen: ngrep -d any -qt -W byline port 5060 Cheers, Daniel On 17/03/2017 06:32, b...@wtf.com.au wrote: Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@ Contact:: sip:5546@:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018@10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp::5060 Methods:: 6815 AOR:: 5...@sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db@10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip::1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp::5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cisco spa502g issues
Hi Daniel, I have attached the gzipped ngrep log I created. There are a number of calls between two of the phones in question, all with out audio working. The phones work fine when calling a different model, or when receiving/make external calls. Ive had a look as I changed the ip address and domain, but really dont know what I am looking for. Cheers, Ben. On 17/03/17 18:04, Daniel-Constantin Mierla wrote: Hello, maybe the new phones do STUN and or the ALG breaks somehow the signaling. You should send the ngrep output taken on sip server for such a call in order to be able to analyze what can happen: ngrep -d any -qt -W byline port 5060 Cheers, Daniel On 17/03/2017 06:32, b...@wtf.com.au wrote: Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@ Contact:: sip:5546@:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018@10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp::5060 Methods:: 6815 AOR:: 5...@sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db@10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip::1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp::5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ngrep2.log.gz Description: application/gzip ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] source port used for forwarded requests
Hi, My Kamailio server listens on multiple IPs - say IP1, IP2. If a request arrives on one IP, Kamailio uses the same IP as the source address when forwarding, which is good. But is there any way a UA could generate a request to IP1 that gets forwarded by Kamailio with source address IP2 ? The reason I ask is because Asterisk is setup to trust all calls arriving from IP2 but not IP1. Thanks Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] source port used for forwarded requests
On Fri, Dec 16, 2011 at 12:26 PM, Alex Balashov wrote: > On 12/15/2011 06:25 PM, Ben WIlliams wrote: > >> Hi, >> My Kamailio server listens on multiple IPs - say IP1, IP2. If a >> request arrives on one IP, Kamailio uses the same IP as the source >> address when forwarding, which is good. But is there any way a UA >> could generate a request to IP1 that gets forwarded by Kamailio with >> source address IP2 ? The reason I ask is because Asterisk is setup to >> trust all calls arriving from IP2 but not IP1. > > > Yes, this is the function of force_send_socket() and/or $fs. Thanks Alex. I didn't actually want to change the socket in my kamailio script. Just checking that there is no way the SIP message itself can specify which source IP is used when forwarding. Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] dialog module - set_dlg_profile won't work
Hi,
can anyone see why this code wouldn't work? In the log is shows this
message after set_dlg_profile called - dialog [dlg_hash.c:523]: no
dialog callid='...' found
Dialogs are being saved to dialog table ok but nothing is in dialog_vars table.
modparam("dialog", "dlg_flag", DLGFLAG)
modparam("dialog", "hash_size", 128)
modparam("dialog", "default_timeout", 14400)
modparam("dialog", "dlg_match_mode", 0)
modparam("dialog", "table_name", "dialog")
modparam("dialog", "db_url", DBURL)
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_update_period", 10)
modparam("dialog", "profiles_with_value", "caller ; allcalls")
...
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
setflag(DLGFLAG); ## add to dialog list
$var(callcnt) = 0;
get_profile_size("caller", "$fu", "$var(callcnt)");
xlog("$fu has $var(callcnt) concurrent calls\n");
if ($var(callcnt) >= 2) {
sl_send_reply("503", "Simultaneous calls limit reached");
exit;
}
set_dlg_profile("caller","$fu");
}
--
Thanks
Ben
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Re: [SR-Users] dialog module - set_dlg_profile won't work
Thanks Carsten, it works now.
Ben
On Thu, Jan 5, 2012 at 9:41 PM, Carsten Bock wrote:
> Hi Ben,
>
> your example is not working, since you set the flag for usage of the
> dialog module, but the dialog has not been created yet. If you add
> "dlg_manage()" after the setting of the according flags, you will get
> a proper value from the get_profile_size()
> The dialog_vars Table is for per dialog-variables, e.g. values you
> want to save additionally to your dialog, e.g.:
> $dlg_var(billing_server) = 'some server name';
> You can then retrieve these values in subsequent requests (e.g.
> Re-INVITES or BYE-Messages).
>
> That's why the dialog_vars table is empty (and in your example it should be).
>
> Carsten
>
> 2012/1/5 Ben WIlliams :
>> Hi,
>> can anyone see why this code wouldn't work? In the log is shows this
>> message after set_dlg_profile called - dialog [dlg_hash.c:523]: no
>> dialog callid='...' found
>> Dialogs are being saved to dialog table ok but nothing is in dialog_vars
>> table.
>>
>> modparam("dialog", "dlg_flag", DLGFLAG)
>> modparam("dialog", "hash_size", 128)
>> modparam("dialog", "default_timeout", 14400)
>> modparam("dialog", "dlg_match_mode", 0)
>> modparam("dialog", "table_name", "dialog")
>> modparam("dialog", "db_url", DBURL)
>> modparam("dialog", "db_mode", 2)
>> modparam("dialog", "db_update_period", 10)
>> modparam("dialog", "profiles_with_value", "caller ; allcalls")
>> ...
>> if (is_method("INVITE")) {
>> setflag(FLT_ACC); # do accounting
>> setflag(DLGFLAG); ## add to dialog list
>>
>> $var(callcnt) = 0;
>> get_profile_size("caller", "$fu", "$var(callcnt)");
>> xlog("$fu has $var(callcnt) concurrent calls\n");
>> if ($var(callcnt) >= 2) {
>> sl_send_reply("503", "Simultaneous calls limit reached");
>> exit;
>> }
>> set_dlg_profile("caller","$fu");
>> }
>>
>> --
>> Thanks
>> Ben
>>
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>
>
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Schomburgstr. 80
> D-22767 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Mobile +49 179 2021244
> Office +49 40 34927219
> Fax +49 40 34927220
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
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Re: [SR-Users] Hashed Passwords
Is it possible to make Siremis store passwords in plaintext? I've just
converted to hashed passwords as well and
kamctlrc has STORE_PLAINTEXT_PW=0 but Siremis is still storing
passwords in plaintext.
On Wed, Jan 4, 2012 at 12:28 AM, Ali Jawad wrote:
> Did trigger an email to fast this time, for people who might get here
> using Google
>
> modparam("auth_db", "calculate_ha1", yes)
> should be
> modparam("auth_db", "calculate_ha1", 0)
>
> On Tue, Jan 3, 2012 at 1:03 PM, Ali Jawad wrote:
>> Hi All
>> We are using plain text passwords for authentication, for the obvious
>> reasons we want to change to hashed passwords so first I executed the
>> following on the SQL server :
>>
>> update subscriber set ha1 = md5(concat(username, ':', domain, ':',
>> password)), ha1b = md5(concat(username, '@', domain, ':', domain, ':',
>> password))
>>
>> The related config is :
>>
>>
>>
>> # - auth_db params -
>> #!ifdef WITH_AUTH
>> modparam("auth_db", "db_url", DBURL)
>> modparam("auth_db", "calculate_ha1", yes)
>> modparam("auth_db", "password_column", "ha1")
>> modparam("auth_db", "load_credentials", "")
>> modparam("auth_db", "use_domain", MULTIDOMAIN)
>>
>> and
>>
>> # - auth_db params -
>> #!ifdef WITH_AUTH
>> modparam("auth_db", "db_url", DBURL)
>> modparam("auth_db", "calculate_ha1", yes)
>> modparam("auth_db", "password_column", "ha1")
>> modparam("auth_db", "load_credentials", "")
>> modparam("auth_db", "use_domain", 1)
>>
>>
>> But I keep getting unauthorized, what did I miss here.
>>
>> Regards
>
>
>
> --
> Ali Jawad
> Information Systems Manager
> Splendor Telecom (www.splendor.net)
> Beirut, Lebanon
> Phone: +9611373725/ext 116
> FAX: +9611375554
>
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[SR-Users] How to do request URI rewrites using database tables?
Hi, can someone please recommend the most appropriate modules to rewrite R-URIs based on a database lookup table? I've read the documentation for lcr and carrierroute but not sure if they can do this. In most cases it will be a simple R-URI rewrite but I also need to rewrite based on From user. ie R-URI match From match new R-URI === == = *9...@example.com 6[0-9]@example.com *97@10.0.0.1 *9...@example.com 7[0-9]@example.com *97@10.0.0.2 6[0-9]@example.com match any rewrite domain to 10.0.0.1 a...@example.com match any d...@example.com Thanks Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to do request URI rewrites using database tables?
I've managed to get dbaliases and lcr to do most of this. But the lcr prefix does not allow regular expressions. Is there any other module that allows you to store in the database a regular expression rewrite rule? On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams wrote: > Hi, can someone please recommend the most appropriate modules to > rewrite R-URIs based on a database lookup table? I've read the > documentation for lcr and carrierroute but not sure if they can do > this. > > In most cases it will be a simple R-URI rewrite but I also need to > rewrite based on From user. > > ie > > R-URI match From match new R-URI > === == = > *9...@example.com 6[0-9]@example.com *97@10.0.0.1 > *9...@example.com 7[0-9]@example.com *97@10.0.0.2 > 6[0-9]@example.com match any rewrite domain to 10.0.0.1 > a...@example.com match any d...@example.com > > Thanks > Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to do request URI rewrites using database tables
Thanks Javi, I'm trying to debug my dialplan but this command does not work : sercmd dp_translate 1 "abcdxyz" It is kamailio 3.2.2 Also, are there any resources that explain how dialplan works other than the supplied documentation? Thanks Ben On Fri, Feb 17, 2012 at 7:36 PM, Javier Gallart wrote: > Hello Ben > > dialplan module should help > you: http://kamailio.org/docs/modules/3.2.x/modules/dialplan.html > > Regards > > > Javi >> >> -- >> >> Message: 5 >> Date: Fri, 17 Feb 2012 12:49:04 +1300 >> From: Ben WIlliams >> Subject: Re: [SR-Users] How to do request URI rewrites using database >> tables? >> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >> Users Mailing List" >> Message-ID: >> >> >> Content-Type: text/plain; charset=ISO-8859-1 >> >> I've managed to get dbaliases and lcr to do most of this. But the lcr >> prefix does not allow regular expressions. Is there any other module >> that allows you to store in the database a regular expression rewrite >> rule? >> >> On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams >> wrote: >> > Hi, can someone please recommend the most appropriate modules to >> > rewrite R-URIs based on a database lookup table? I've read the >> > documentation for lcr and carrierroute but not sure if they can do >> > this. >> > >> > In most cases it will be a simple R-URI rewrite but I also need to >> > rewrite based on From user. >> > >> > ie >> > >> > R-URI match ? ? ? ? ? ? From match ? ? ? ? ? ? ? ? ?new R-URI >> > === ? ? ? ? ? ? == ? ? ? ? ? ? ? ? ?= >> > *9...@example.com ? ? ? ? 6[0-9]@example.com ? ? ? ? ?*97@10.0.0.1 >> > *9...@example.com ? ? ? ? 7[0-9]@example.com ? ? ? ? ?*97@10.0.0.2 >> > 6[0-9]@example.com ? ? ?match any ? ? ? ? ? ? ? ? ? rewrite domain to >> > 10.0.0.1 >> > a...@example.com ? ? ? ? match any ? ? ? ? ? ? ? ? ? d...@example.com >> > >> > Thanks >> > Ben >> >> >> > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] lcr from_uri not matching
Has anyone experienced this problem where lcr only works when from_uri is null. When I change it to .* the match fails. Thanks Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] lcr from_uri not matching
Thanks for checking. It was the only rule I had and it was working but only when from_uri was null. On Tue, Feb 28, 2012 at 2:59 PM, Juha Heinanen wrote: > Ben WIlliams writes: > >> Has anyone experienced this problem where lcr only works when from_uri >> is null. When I change it to .* the match fails. > > i just tested with latest master the case where from uri is .* and it > worked as expected. check that the rule is enabled in lcr_rule table > and that a gw is defined for the rule in rule_target table. > > -- juha > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] lcr from_uri not matching
Its 3.2.2 from http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-6/ On Tue, Feb 28, 2012 at 3:37 PM, Juha Heinanen wrote: > Ben WIlliams writes: > >> Thanks for checking. It was the only rule I had and it was working but >> only when from_uri was null. > > which version of kamailio are you using? > > -- juha > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] lcr from_uri not matching
Hi Juha, here is the dump, prefix 8 should use gateway number 2. Its not a big issue, I've managed to do it without LCR now. INSERT INTO `lcr_gw` VALUES (1,1,'chc','1.1.1.1',NULL,NULL,NULL,1,1,NULL,NULL,NULL,1,NULL),(2,1,'ak','2.2.2.2',NULL,NULL,NULL,1,1,NULL,NULL,NULL,1,NULL); INSERT INTO `lcr_rule` VALUES (1,1,'8','.*',1,1); INSERT INTO `lcr_rule_target` VALUES (1,1,1,2,1,1); Thanks Ben On Wed, Feb 29, 2012 at 3:20 PM, Juha Heinanen wrote: > Ben WIlliams writes: > >> Its 3.2.2 from >> http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-6/ > > can you post dump of your lcr tables and i'll give a try with them? > > -- juha > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] myself variable
Hi, I am an absolute beginniner learning SIP and Kamailio to setup Kamailio as the Internet proxy for our Asterisk PBX. My question is how to list the contents of the myself variable from the command line? My Kamailio server displays a list of Aliases on startup. Are these the contents of the myself variable? Thanks Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] sipcapture module with wireshark
Ali, If you don't mind me asking, how are you using the data you capture with sipcapture? Are you using a Web gui of sorts to view the SIP information? I only ask, because various SIP analysis GUI's, such as HOMER, pcapture etc. have functions built into them that allow you to analyse SIP dialogs, and then download them as PCAPs. Kind Regards, ----- Ben Bliss Network Systems Administrator Entanet International Ltd T: 0333 101 9229 F: 0333 101 0601 E: supp...@enta.net W: http://www.enta.net W: http://opinion.enta.net This e-mail and any attachments transmitted with it, including replies and forwarded copies subsequently transmitted, contains information which may be confidential and which may also be legally privileged. The content of this e-mail is for the exclusive use of the individual(s) or entity named above, if you are not the intended recipient, please note that any form of distribution, copying or use of this e-mail or the information in it is strictly prohibited and may be unlawful. If you have received this e-mail in error please reply to the sender at Entanet International Ltd and then delete the e-mail immediately. Any views or opinions presented are solely those of the author, and may not necessarily represent those of Entanet International Ltd. Entanet international Ltd is registered in England and Wales, registered no 3274237, and its registered office is Stafford Park 6, Telford, Shropshire, TF3 3AT. registered VAT number GB 184 5136 01. On 14/04/2016 4:54 PM, Ali Taher wrote: Hello, I’m using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I’m wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Regards, Ali ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Registrar module and local paths
Hello everyone. I'm having a problem with registrations and Path and hope
that someone can help.
I have a Kamailio instance which is both a WebSocket proxy and a registrar
which has the following behaviour:
* A REGISTER with a Path header already on it gets recorded in the location
table.
* I use add_path() on all incoming REGISTERs, and a REGISTER relayed to
another registrar reaches that registrar with a Path header added
appropriately.
Unfortunately, save("location") ignores the path added by add_path(). I'd
prefer to keep the proxy and registrar co-located for the short term. Does
anyone know if/how I might be able to have the Path header properly stored
in the case that it was added by add_path()?
Regards,
Ben Langfeld
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Re: [SR-Users] Registrar module and local paths
> Em 15/10/2014, às 07:18, Daniel-Constantin Mierla
> escreveu:
>
> Hello,
>
>> On 14/10/14 23:56, Ben Langfeld wrote:
>> Hello everyone. I'm having a problem with registrations and Path and hope
>> that someone can help.
>>
>> I have a Kamailio instance which is both a WebSocket proxy and a registrar
>> which has the following behaviour:
>>
>> * A REGISTER with a Path header already on it gets recorded in the location
>> table.
>>
>> * I use add_path() on all incoming REGISTERs, and a REGISTER relayed to
>> another registrar reaches that registrar with a Path header added
>> appropriately.
>>
>> Unfortunately, save("location") ignores the path added by add_path(). I'd
>> prefer to keep the proxy and registrar co-located for the short term. Does
>> anyone know if/how I might be able to have the Path header properly stored
>> in the case that it was added by add_path()?
> if you have the registrar as first hop from client, then you don't need to
> save path. save(location) and lookup(location) shout take care of everything.
> Just be sure you set the nat flag and use fix_nated_register() also for
> traffic coming via websocket.
Unfortunately it's not that simple. I have two or more instances of this
combined proxy/registrar and usrloc in dbmode 3 so they can share location data
without SIP level replication. A web socket UA on one proxy would need to use
the path to reach a we socket UA on the other proxy, else it looks locally for
the socket and fails.
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Re: [SR-Users] Registrar module and local paths
On 15 October 2014 10:33, Daniel-Constantin Mierla
wrote:
>
> On 15/10/14 14:19, Ben Langfeld wrote:
> >> Em 15/10/2014, às 07:18, Daniel-Constantin Mierla
> escreveu:
> >>
> >> Hello,
> >>
> >>> On 14/10/14 23:56, Ben Langfeld wrote:
> >>> Hello everyone. I'm having a problem with registrations and Path and
> hope that someone can help.
> >>>
> >>> I have a Kamailio instance which is both a WebSocket proxy and a
> registrar which has the following behaviour:
> >>>
> >>> * A REGISTER with a Path header already on it gets recorded in the
> location table.
> >>>
> >>> * I use add_path() on all incoming REGISTERs, and a REGISTER relayed
> to another registrar reaches that registrar with a Path header added
> appropriately.
> >>>
> >>> Unfortunately, save("location") ignores the path added by add_path().
> I'd prefer to keep the proxy and registrar co-located for the short term.
> Does anyone know if/how I might be able to have the Path header properly
> stored in the case that it was added by add_path()?
> >> if you have the registrar as first hop from client, then you don't need
> to save path. save(location) and lookup(location) shout take care of
> everything. Just be sure you set the nat flag and use fix_nated_register()
> also for traffic coming via websocket.
> > Unfortunately it's not that simple. I have two or more instances of this
> combined proxy/registrar and usrloc in dbmode 3 so they can share location
> data without SIP level replication. A web socket UA on one proxy would need
> to use the path to reach a we socket UA on the other proxy, else it looks
> locally for the socket and fails.
> >
> But then, the instance handling the REGISTER will have to do a loop or
> other tricks in config to avoid it.
>
> See msg_apply_changes() from textopsx to get the path header visible.
Thanks for the reference Daniel. I gave this a try and it results in the
following:
Oct 15 14:03:06 nyubachibo /usr/sbin/kamailio[21350]: DEBUG:
[parser/parse_uri.c:1284]: parse_uri(): parse_uri: bad host in uri (error
at char ; in state 4) parsed: (37)
/ (43)
Oct 15 14:03:06 nyubachibo /usr/sbin/kamailio[21350]: ERROR: registrar
[path.c:84]: build_path_vector(): failed to parse the first Path URI
I figure at this point it may be simpler to separate the registrar and the
proxy rather than attempt to debug this further, though if you have any
other suggestions to avoid that I'd love to hear them.
Thanks for your help so far, it's much appreciated :)
Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Registrar module and local paths
On 15 October 2014 11:32, Juha Heinanen wrote: > Ben Langfeld writes: > > > I figure at this point it may be simpler to separate the registrar and > the > > proxy rather than attempt to debug this further, though if you have any > > other suggestions to avoid that I'd love to hear them. > > one possibility is that both of your combined proxy/registrars have > their own location tables and you forward registrations from one to the > other. > The problem with that is horizontal scalability brings noise. If I have 10 of these things, the SIP replication alone would be flooding the network. > -- juha > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Registrar module and local paths
To follow up, separately deployed WebSocket proxies and registrars (sharing usrloc DB w/ db_mode 3) are working nicely now. Perhaps I can optimise the shared location data further, but that's a good enough start for me :) Thanks for the input from everyone. On 15 October 2014 11:47, Frank Carmickle wrote: > > On Oct 15, 2014, at 10:40 AM, Ben Langfeld wrote: > > On 15 October 2014 11:32, Juha Heinanen wrote: > >> Ben Langfeld writes: >> >> > I figure at this point it may be simpler to separate the registrar and >> the >> > proxy rather than attempt to debug this further, though if you have any >> > other suggestions to avoid that I'd love to hear them. >> >> one possibility is that both of your combined proxy/registrars have >> their own location tables and you forward registrations from one to the >> other. >> > > The problem with that is horizontal scalability brings noise. If I have 10 > of these things, the SIP replication alone would be flooding the network. > > > If you are running in an environment where you can use multicast it might > be an option for you. Multicast the registrations from the edge proxy to > the registrar cluster. If not maybe you can get the registrars to > replicate to each other on a separate interface from the interface facing > the edge proxy. > > > --FC > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Configuring TLS and WSS with Kamailio
Out of curiosity, why don't you use the pre-built packaged binaries instead
of building from source?
On 17 October 2014 19:53, Kamrul Khan wrote:
> Hi,
>
>
> Im trying to connect my WebRTC clietn to kamailio via WSS. I
> successfully connected it via WS but having trouble connecting it via WSS.
> My first issue is I get error messages while i try to compile TLS
> module(console log in the end of the document). But, still it creates
> tls.so file. So i copied the tls.so to my kamailio modules directory and
> then updated my kamailio configuration as below:
>
>
> #!define WITH_TLS
>
> .
>
> .
>
> listen=tcp:127.0.0.1:5061
>
> listen=tcp:192.168.146.133:5061
>
> .
>
> .
>
> #!ifdef WITH_TLS
>
> enable_tls=yes
>
> #!endif
>
> .
>
> .
>
> #!ifdef WITH_TLS
>
> # - tls params -
>
> modparam("tls", "config", "/usr/local/kamailio-devel/etc/kamailio/tls.cfg")
>
> #!endif
>
>
> Now, according to this mailing list,
> http://lists.sip-router.org/pipermail/sr-users/2013-March/077182.html : “*when
> tls module is installed, a self signed pair of certificate-private key is
> generated in /usr/local/etc/kamailio”*
>
>
> In my case, I dont even have the /usr/local/etc/kamailio directory. So,
> I followed
> http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates to
> create my certificate and key. And updated my tsl.cfg, this is how my
> tsl.cfg looks like:
>
>
> [server:192.168.146.133:5061]
>
> method = TLSv1
>
> verify_certificate = no
>
> require_certificate = no
>
>
> private_key = /etc/certs/sip.192.168.146.133/key.pem
>
> certificate = /etc/certs/sip.192.168.146.133/cert.pem
>
> ca_list = /etc/certs/demoCA/cert.pem
>
>
> [client:192.168.146.133:5061]
>
> verify_certificate = yes
>
> require_certificate = yes
>
>
>
> Then I restarted my kamailio server. And i get the following errors in
> my kamailio log:
>
>
> Im not sure what wrong im doing. Please help me:
>
>
> Oct 17 15:44:50 ubuntu kamailio: INFO: tls [tls_init.c:385]:
> init_tls_compression(): tls: init_tls: disabling compression...
>
> Oct 17 15:44:50 ubuntu kamailio: WARNING: [socket_info.c:1397]:
> fix_hostname(): WARNING: fix_hostname: could not rev. resolve
> 192.168.146.133
>
> Oct 17 15:44:50 ubuntu kamailio: message repeated 2 times: [ WARNING:
> [socket_info.c:1397]: fix_hostname(): WARNING: fix_hostname: could
> not rev. resolve 192.168.146.133]
>
> Oct 17 15:44:50 ubuntu kamailio: INFO: [tcp_main.c:4836]:
> init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)
>
> Oct 17 15:44:50 ubuntu kamailio: WARNING: [daemonize.c:352]:
> daemonize(): pid file contains old pid, replacing pid
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: utils [utils.c:288]: mod_init(): forward functionality disabled
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: utils [utils.c:197]: pres_db_init(): xcap_auth_status function is
> disabled
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> ERROR: tls [tls_init.c:668]: tls_check_sockets(): TLSs<
> 192.168.146.133:5061>: No listening socket found
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> ERROR: [sr_module.c:970]: init_mod(): init_mod(): Error while
> initializing module tls
> (/usr/local/kamailio-devel/lib64/kamailio/modules/tls.so)
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: :
> tls [tls_locking.c:103]: locking_f(): BUG: tls: locking_f (callback):
> invalid lock number: 30 (range 0 - 0), called from eng_table.c:227
>
> Oct 17 15:44:51 ubuntu kamailio: ERROR: [daemonize.c:307]:
> daemonize(): Main process exited before writing to pipe
>
>
> tls module compile log
>
>
> xxx@ubuntu:/usr/local/src/kamailio-4.1/kamailio$ sudo make -C modules/tls
> make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls'
> make: Leaving directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls'
> make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls'
> CC (gcc) [M tls.so] tls_bio.o
> CC (gcc) [M tls.so] tls_cfg.o
> CC (gcc) [M tls.so] tls_config.o
> CC (gcc) [M tls.so] tls_ct_wrq.o
> CC (gcc) [M tls.so] tls_domain.o
> In file included from tls_domain.c:39:0:
> tls_domain.c: In function âload_certâ:
> tls_util.h:52:6: warning: variable âretâ set but not used
> [-Wunused-but-set-variable]
> int ret; \
> ^
> tls_domain.c:506:4: note: in expansion of macro âTLS_ERRâ
> TLS_ERR("lo
Re: [SR-Users] Configuring TLS and WSS with Kamailio
These are distributed via package repositories. I'm not sure what you mean
by "standard installation" - you mean a source tarball?
Debian: http://www.kamailio.org/wiki/packages/debs
RH variants: http://www.kamailio.org/wiki/packages/rpms
On 17 October 2014 20:08, Kamrul Khan wrote:
> I dint find any in my installation directory. probably it doesn't come
> with standard installation unless you explicitly mention for it... not sure
> though
>
> --
> Date: Fri, 17 Oct 2014 19:56:50 -0300
> From: b...@langfeld.co.uk
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] Configuring TLS and WSS with Kamailio
>
>
> Out of curiosity, why don't you use the pre-built packaged binaries
> instead of building from source?
>
> On 17 October 2014 19:53, Kamrul Khan wrote:
>
> Hi,
>
>
> Im trying to connect my WebRTC clietn to kamailio via WSS. I
> successfully connected it via WS but having trouble connecting it via WSS.
> My first issue is I get error messages while i try to compile TLS
> module(console log in the end of the document). But, still it creates
> tls.so file. So i copied the tls.so to my kamailio modules directory and
> then updated my kamailio configuration as below:
>
>
> #!define WITH_TLS
>
> .
>
> .
>
> listen=tcp:127.0.0.1:5061
>
> listen=tcp:192.168.146.133:5061
>
> .
>
> .
>
> #!ifdef WITH_TLS
>
> enable_tls=yes
>
> #!endif
>
> .
>
> .
>
> #!ifdef WITH_TLS
>
> # - tls params -
>
> modparam("tls", "config", "/usr/local/kamailio-devel/etc/kamailio/tls.cfg")
>
> #!endif
>
>
> Now, according to this mailing list,
> http://lists.sip-router.org/pipermail/sr-users/2013-March/077182.html : “*when
> tls module is installed, a self signed pair of certificate-private key is
> generated in /usr/local/etc/kamailio”*
>
>
> In my case, I dont even have the /usr/local/etc/kamailio directory. So,
> I followed
> http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates to
> create my certificate and key. And updated my tsl.cfg, this is how my
> tsl.cfg looks like:
>
>
> [server:192.168.146.133:5061]
>
> method = TLSv1
>
> verify_certificate = no
>
> require_certificate = no
>
>
> private_key = /etc/certs/sip.192.168.146.133/key.pem
>
> certificate = /etc/certs/sip.192.168.146.133/cert.pem
>
> ca_list = /etc/certs/demoCA/cert.pem
>
>
> [client:192.168.146.133:5061]
>
> verify_certificate = yes
>
> require_certificate = yes
>
>
>
> Then I restarted my kamailio server. And i get the following errors in
> my kamailio log:
>
>
> Im not sure what wrong im doing. Please help me:
>
>
> Oct 17 15:44:50 ubuntu kamailio: INFO: tls [tls_init.c:385]:
> init_tls_compression(): tls: init_tls: disabling compression...
>
> Oct 17 15:44:50 ubuntu kamailio: WARNING: [socket_info.c:1397]:
> fix_hostname(): WARNING: fix_hostname: could not rev. resolve
> 192.168.146.133
>
> Oct 17 15:44:50 ubuntu kamailio: message repeated 2 times: [ WARNING:
> [socket_info.c:1397]: fix_hostname(): WARNING: fix_hostname: could
> not rev. resolve 192.168.146.133]
>
> Oct 17 15:44:50 ubuntu kamailio: INFO: [tcp_main.c:4836]:
> init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)
>
> Oct 17 15:44:50 ubuntu kamailio: WARNING: [daemonize.c:352]:
> daemonize(): pid file contains old pid, replacing pid
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: utils [utils.c:288]: mod_init(): forward functionality disabled
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> INFO: utils [utils.c:197]: pres_db_init(): xcap_auth_status function is
> disabled
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> ERROR: tls [tls_init.c:668]: tls_check_sockets(): TLSs<
> 192.168.146.133:5061>: No listening socket found
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]:
> ERROR: [sr_module.c:970]: init_mod(): init_mod(): Error while
> initializing module tls
> (/usr/local/kamailio-devel/lib64/kamailio/modules/tls.so)
>
> Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: :
> tls [tls_locking.c:103]: locking_f(): BUG: tls: locking_f (callback):
> invalid lock number: 30 (range 0 - 0), called from eng_table.c:227
>
> Oct 17 15:44:51 ubuntu kamailio: ERROR: [daemonize.c:307]:
> daemonize(): Main process exited before writing to pipe
>
>
> tls module compile log
>
>
> xxx@ubuntu:/usr/local/src/kamailio-4.1/kamailio$ sudo make -C modules/tls
> make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls'
> make: Le
Re: [SR-Users] Setting up web sockets and ephemeral auth
The important part is this:
wrong char [U/85]
in [$hdr(
Upgrade)] at [5 (5)]
You have a unicode newline character after the opening parenthesis. You
should remove this and any other instances of weird characters which have
become inserted in your config.
On 24 October 2014 18:50, Nolan Darilek wrote:
> Having a hard time piecing together lots of pieces out of context.
> Here's what I want:
>
> 1. User logs in to my web app via its own authentication mechanisms.
> 2. Via an authenticated REST endpoint, my web app connects to the
> ephemeral auth service and returns a temporary username/password to the
> client.
> 3. The client connects to a web socket SIP interface, authenticating
> with the username/password it was given.
> 4. I'd also like to support authentication via standard SIP client using
> the user's default username/password, but that's a bit down the road still.
>
> Here is my kamailio-local.cfg. I'm using the web socket example copied
> from the docs, with a little cleanup and a few optional checks removed
> for now.
>
> loadmodule "db_mongodb.so"
>
> #!define DBURL "mongodb://mongo/perceptron"
>
> #!define WITH_AUTH
>
> #!define WITH_NAT
>
> loadmodule "xhttp.so"
>
> #modparam("xhttp", "url_match", "^/sip/")
>
> tcp_accept_no_cl=yes
>
> loadmodule "msrp.so" # Only required if using MSRP over WebSockets
> loadmodule "websocket.so"
>
> loadmodule "auth_ephemeral.so"
>
> modparam("auth_ephemeral", "secret", "Kamailio rulez!!11")
>
> event_route[xhttp:request] {
> set_reply_close();
> set_reply_no_connect();
>
> # xlog("L_DBG", "HTTP Request Received\n");
>
> if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" &&
> $rm=~ "GET" ) {
> # Validate Host - make sure the client is using the correct
> # alias for WebSockets
> if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) {
> # xlog("L_WARN", "Bad host $hdr(Host)\n");
> xhttp_reply("403", "Forbidden", "", "");
> exit;
> }
>
> if (ws_handle_handshake()) {
> exit;
> }
> }
>
> xhttp_reply("404", "Not found", "", "");
>
> }
>
>
> When I attempt to validate this, I get:
>
> 0(1) ERROR: [pvapi.c:790]: pv_parse_spec2(): error searching
> pvar "hdr"
> 0(1) ERROR: [pvapi.c:994]: pv_parse_spec2(): wrong char [U/85]
> in [$hdr(
> Upgrade)] at [5 (5)]
> 0(1) : [cfg.y:3436]: yyerror_at(): parse error in config file
> /usr/local
> /etc/kamailio/kamailio-local.cfg, line 28, column 7-19: Can't get from
> cache: $h
> dr(Upgrade)
> ERROR: bad config file (1 errors)
>
> I don't know what that means. I've copied the config directly from the
> website.
>
> I tried quoting "Upgrade" and such in the $hdr lines, but that gives me:
>
> 0(1) ERROR: [pvapi.c:790]: pv_parse_spec2(): error searching
> pvar "hdr"
> 0(1) ERROR: [pvapi.c:994]: pv_parse_spec2(): wrong char ["/34]
> in [$hdr(
> "Upgrade")] at [5 (5)]
> 0(1) : [cfg.y:3436]: yyerror_at(): parse error in config file
> /usr/local
> /etc/kamailio/kamailio-local.cfg, line 28, column 7-21: Can't get from
> cache: $h
> dr("Upgrade")
> ERROR: bad config file (1 errors)
>
>
> What am I missing here?
>
> I also had to comment out the xlog lines because those gave me errors.
> Do I need another module loaded to use xlog?
>
> Finally, does anyone have a working, complete example of SIP over web
> sockets with ephemeral auth integrated? I'm not sure if I should be
> checking the ephemeral auth credentials in the web sockets code, the SIP
> routes or somewhere else. I'm also not sure how I should set up
> authentication such that users can log in with both their standard
> credentials or those retrieved via the ephemeral auth module.
>
> Thanks.
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] Error-While-Running-icscf.cfg
You would have to provide your config file. Also, it's bad form to email project maintainers directly; they read the list and will respond here if and when they can. On 27 October 2014 04:21, Anil Kumar wrote: > Hi All, > > > Please can anyone help me to solve this error, occurred while running > icscf module . > > 0(4053) : [cfg.y:3406]: yyerror_at(): parse error in config file > /etc/kamailio-icscf/icscf.cfg, from line 52, column 1 to line 53, column 0: > syntax error > ERROR: bad config file (1 errors) > > > Thanks > Anil > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio as pass through proxy?
You could quite easily just do this with iptables. See http://askubuntu.com/questions/320121/simple-port-forwarding for ideas. On 27 November 2014 at 05:29, Rizwan Khan wrote: > Hi All, > > What we want to do is to by-pass the restrictions imposed by different > ISPs etc. which normally block based on the port 5060 > > The SIP server(including SBC) is a commercial one running on port 5060, > that we cannot change. It is already taking care of the NAT issues. > > We want to use Kamailio as the pass-thru proxy in front of the SBC just to > receive the request on a random port, and then forward the request to the > SBC internally being on the same private network. > > What would be the best way to do it? > > Any help will be highly appreciated. > > > Rizwan Khan > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] WebRTC + Kamailio as SIP-proxy + rtpproxy(?) as media-proxy
The main thing you need to look out for is that your registrar supports the Path and Outbound specifications in order to correctly route INVITEs to your WebSocket clients via the edge proxy. I'm in a situation right now where I'm having some difficulty getting a Kamailio WebSocket edge proxy playing nice with an Asterisk 1.4 registrar, which doesn't support those specs. If anyone has any tips, I'd love to hear them. On 27 November 2014 at 10:45, Camille Oudot wrote: > Hi, > > > a) Can kamailio be used as sip-proxy while using WebRTC based UA > > calling to plain UAC/WebRTC based UAC ? > > Yes, kamailio can do SIP over websocket, so all you need is a > javascript SIP stack (e.g. JsSIP, jain-sip JS, ...) on your WebRTC > enabled client. > > > b) What to use for media proxying (this really baffles me..) rtpproxy > > or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation > > between them anywhere? > > you will need to be able to translate WebRTC RTP (RTP/SAVPF) to other > RTP profiles like RTP/AVP. Only rtpengine can do this (note that > mediaproxy-ng is the old name for rtpengine). > > > c) I am not behind NAT and do not want secure web-sockets, so any > > sample config I can refer to ? > > If you familiar with kamailio cfg scripting you can try to start > something from scratch (building a simple proxy is quite > straightforward). Otherwise i don't know any example file that does all > you need. > > See examples/websocket.cfg for websocket handling. You can disable the > registrar and the NAT stuff in it if you don't need them. > > Cheers, > > -- > Camille > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Registrar path support on lookup
I'm facing a problem which I believe may be a bug in the registrar module's lookup. When a registration comes with multiple Path headers, these are recorded in the location table as comma separated. When the lookup is performed and these are used to construct the Route header(s) on an INVITE, this comma-separated list is not split into multiple Route headers, but instead included verbatim in a single header. Here comes my theory for how this breaks my scenario: When the INVITE then reaches the first hop (first Path header, also Kamailio), relay() sees that the Route header is itself and/or sees that there is only one Route header. The observed result is that the first hop then sends the INVITE directly to the Contact, instead of via the second hop. An example of the situation I'm facing is shown in https://gist.github.com/benlangfeld/b374a0ce0bdb6bdc35e7. Is there a particular reason the combined path is not split on lookup? Is this a bug / oversight? Am I crazy? Thanks! Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Registrar path support on lookup
In that case the issue would have to be with the Kamailio-based edge proxy which is not routing based on the second Route in the list. Is that a known issue? I can provide more details if you think I'm on the right track. On 4 December 2014 at 10:09, Daniel-Constantin Mierla wrote: > Hello, > > having more record-route addresses in the same header, separated by comma, > it is valid. Other headers (Via, Record-Route, Supported, Required ...) can > have same format, it is a matter of UAC how it builds them, each with a > header name or all under same header name. > > Cheers, > Daniel > > > On 04/12/14 12:24, Ben Langfeld wrote: > > I'm facing a problem which I believe may be a bug in the registrar > module's lookup. When a registration comes with multiple Path headers, > these are recorded in the location table as comma separated. When the > lookup is performed and these are used to construct the Route header(s) on > an INVITE, this comma-separated list is not split into multiple Route > headers, but instead included verbatim in a single header. > > Here comes my theory for how this breaks my scenario: > > When the INVITE then reaches the first hop (first Path header, also > Kamailio), relay() sees that the Route header is itself and/or sees that > there is only one Route header. > > The observed result is that the first hop then sends the INVITE directly > to the Contact, instead of via the second hop. > > An example of the situation I'm facing is shown in > https://gist.github.com/benlangfeld/b374a0ce0bdb6bdc35e7. > > Is there a particular reason the combined path is not split on lookup? > Is this a bug / oversight? Am I crazy? > > Thanks! > Ben > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - > http://www.linkedin.com/in/miconda > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Registrar path support on lookup
Thanks Daniel. Config can be found here: https://gist.github.com/benlangfeld/1b61f41c31129e8f2db3 INVITEs as provided (with a Route with two comma-separated elements) get routed via the Contact header (instead of the second Route element). When I use loose_route() on the INVITE, it goes into a loop in the edge proxy (https://gist.github.com/benlangfeld/99af9045e70a9bb4eeed), even though the IP it selects to route to (.14) is not the same server. Ben On 4 December 2014 at 12:16, Daniel-Constantin Mierla wrote: > You should put the config somewhere for review. Note that default > kamailio.cfg ignores Route headers for initial requests. You have to handle > initial requests with loose_route(). > > Cheers, > Daniel > > > On 04/12/14 13:35, Ben Langfeld wrote: > > In that case the issue would have to be with the Kamailio-based edge proxy > which is not routing based on the second Route in the list. Is that a known > issue? I can provide more details if you think I'm on the right track. > > On 4 December 2014 at 10:09, Daniel-Constantin Mierla > wrote: > >> Hello, >> >> having more record-route addresses in the same header, separated by >> comma, it is valid. Other headers (Via, Record-Route, Supported, Required >> ...) can have same format, it is a matter of UAC how it builds them, each >> with a header name or all under same header name. >> >> Cheers, >> Daniel >> >> >> On 04/12/14 12:24, Ben Langfeld wrote: >> >> I'm facing a problem which I believe may be a bug in the registrar >> module's lookup. When a registration comes with multiple Path headers, >> these are recorded in the location table as comma separated. When the >> lookup is performed and these are used to construct the Route header(s) on >> an INVITE, this comma-separated list is not split into multiple Route >> headers, but instead included verbatim in a single header. >> >> Here comes my theory for how this breaks my scenario: >> >> When the INVITE then reaches the first hop (first Path header, also >> Kamailio), relay() sees that the Route header is itself and/or sees that >> there is only one Route header. >> >> The observed result is that the first hop then sends the INVITE >> directly to the Contact, instead of via the second hop. >> >> An example of the situation I'm facing is shown in >> https://gist.github.com/benlangfeld/b374a0ce0bdb6bdc35e7. >> >> Is there a particular reason the combined path is not split on lookup? >> Is this a bug / oversight? Am I crazy? >> >> Thanks! >> Ben >> >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >> http://www.linkedin.com/in/miconda >> >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - > http://www.linkedin.com/in/miconda > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio not start if redis server is down
What would you like to happen when a service on which you depend is not available? On 22 December 2014 at 15:56, pars3c wrote: > Hi, > > i’m trying to use the ndb_redis connector , but i’ve a problem when redis > server is down and i try to restart kamailio. > > Kamailio in this case not start and release this log: > > > > 2(19590) ERROR: ndb_redis [redis_client.c:112]: redisc_init(): error > communicating with redis server [srv1] (xx/0): Connection refused > > 2(19590) ERROR: ndb_redis [ndb_redis_mod.c:126]: child_init(): failed to > initialize redis connections > > 2(19590) ERROR: [sr_module.c:927]: init_mod_child(): > init_mod_child(): Error while initializing module ndb_redis > (//lib/kamailio/modules/ndb_redis.so) > > > > 30(19622) ERROR: [tcp_main.c:4962]: tcp_init_children(): ERROR: > tcp_main: fork failed: Connection refused > > 31(19624) : [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF > on 11 > > 31(19624) : [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF > on 31 > > 0(19588) ALERT: [main.c:775]: handle_sigs(): child process 19589 > exited normally, status=255 > > 0(19588) ALERT: [main.c:775]: handle_sigs(): child process 19590 > exited normally, status=255 > > > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Active / Active HA.
On 27 December 2014 at 23:44, Carsten Bock wrote: > Hi Mahmoud, > > Thy typical way, to build a Active/Active setup is by using DNS-SRV > records. The two servers get different IPs and you announce both IPs > using DNS to the devices. Most common User-Agents support this > nowadays, so that's kind of easy. It is specified in RFC3263: > "Locating SIP-Servers" (http://tools.ietf.org/html/rfc3263). > > However, when using DNS-SRV records, dialogs and transactions are > still associated with a single server, even though the User-Agent has > an Alternative, in case one server fails. If you want to increase the > HA for this, you can advertise the Domain name instead of an IP > (causes more DNS-Lookups). > Just a word of caution here, there are several SIP stacks which don't handle SRV lookups correctly. I don't mean to discourage you from this, but just to be aware of it. For example, Mobicents only understands SRV record sets which return domain names. If you return IPs, Mobicents will go ahead and look them up as A records and fail. > You can share the dialog state etween two nodes by using the dialog > module in db-only mode (db_mode = 1, see > http://kamailio.org/docs/modules/devel/modules/dialog.html#idp15368320). > You cannot share transactions between two nodes, only dialogs. > > In addition, you can work with virtual IPs with Heartbeat/Pacemaker. > > At a big operator, we've used a combination of the two: Active/Passive > for one site and DNS-SRV to distribute it among different sites. > > Kind regards, > Carsten > > 2014-12-26 19:07 GMT+01:00 Mahmoud Ramadan Ali < > cisco.and.more.b...@gmail.com>: > > Hi Dears, > > I have successfully configured Kamailio HA using Heartbeat and Pacemaker > so > > if one of the two servers should go down the other server will own the > > virtual IP address and take over. > > But i have two questions: > > > > 1.This model is considered to be Active / Passive redundancy so one > server > > will process the SIP signaling until it goes down and i'm wondering is > there > > is any way two achieve Active / Active redundancy and if so how the > > signaling will be handled on the servers so they can be aware of the > > transactions and dialogs traversing the servers ? does the signalling > will > > be replicated or synchronized between the servers or what ?! > > > > 2.What about the servers DB and how they should be designed in a cluster > > mode ? i want to replicate all the DBs of the server to get consistent > user > > registration using the subscriber table for example. > > > > Thanks in advance and Best regards. > > > > ___ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:cars...@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Active / Active HA.
Good point. Mobicents is on my list of projects to which I'm due to contribute something. I'll get on it shortly. On 29 December 2014 at 11:26, Carsten Bock wrote: > Hi Ben, > > that's absolutely for sure! It took us ~2 years to get the DNS-SRV > implementation of major DSL-Modem-Manufacturer right ;-) > Nevertheless, it's the way to go for redundancy. And since Mobicents > is open-source, we can probably simply provide a patch to fix it :-) > > Kind regards, > Carsten > > 2014-12-29 12:13 GMT+01:00 Ben Langfeld : > > On 27 December 2014 at 23:44, Carsten Bock wrote: > >> > >> Hi Mahmoud, > >> > >> Thy typical way, to build a Active/Active setup is by using DNS-SRV > >> records. The two servers get different IPs and you announce both IPs > >> using DNS to the devices. Most common User-Agents support this > >> nowadays, so that's kind of easy. It is specified in RFC3263: > >> "Locating SIP-Servers" (http://tools.ietf.org/html/rfc3263). > >> > >> However, when using DNS-SRV records, dialogs and transactions are > >> still associated with a single server, even though the User-Agent has > >> an Alternative, in case one server fails. If you want to increase the > >> HA for this, you can advertise the Domain name instead of an IP > >> (causes more DNS-Lookups). > > > > > > Just a word of caution here, there are several SIP stacks which don't > handle > > SRV lookups correctly. I don't mean to discourage you from this, but > just to > > be aware of it. For example, Mobicents only understands SRV record sets > > which return domain names. If you return IPs, Mobicents will go ahead and > > look them up as A records and fail. > > > >> > >> You can share the dialog state etween two nodes by using the dialog > >> module in db-only mode (db_mode = 1, see > >> http://kamailio.org/docs/modules/devel/modules/dialog.html#idp15368320 > ). > >> You cannot share transactions between two nodes, only dialogs. > >> > >> In addition, you can work with virtual IPs with Heartbeat/Pacemaker. > >> > >> At a big operator, we've used a combination of the two: Active/Passive > >> for one site and DNS-SRV to distribute it among different sites. > >> > >> Kind regards, > >> Carsten > >> > >> 2014-12-26 19:07 GMT+01:00 Mahmoud Ramadan Ali > >> : > >> > Hi Dears, > >> > I have successfully configured Kamailio HA using Heartbeat and > Pacemaker > >> > so > >> > if one of the two servers should go down the other server will own the > >> > virtual IP address and take over. > >> > But i have two questions: > >> > > >> > 1.This model is considered to be Active / Passive redundancy so one > >> > server > >> > will process the SIP signaling until it goes down and i'm wondering is > >> > there > >> > is any way two achieve Active / Active redundancy and if so how the > >> > signaling will be handled on the servers so they can be aware of the > >> > transactions and dialogs traversing the servers ? does the signalling > >> > will > >> > be replicated or synchronized between the servers or what ?! > >> > > >> > 2.What about the servers DB and how they should be designed in a > cluster > >> > mode ? i want to replicate all the DBs of the server to get consistent > >> > user > >> > registration using the subscriber table for example. > >> > > >> > Thanks in advance and Best regards. > >> > > >> > ___ > >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > >> > sr-users@lists.sip-router.org > >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > >> > >> > >> > >> -- > >> Carsten Bock > >> CEO (Geschäftsführer) > >> > >> ng-voice GmbH > >> Schomburgstr. 80 > >> D-22767 Hamburg / Germany > >> > >> http://www.ng-voice.com > >> mailto:cars...@ng-voice.com > >> > >> Office +49 40 5247593-0 > >> Fax +49 40 5247593-99 > >> > >> Sitz der Gesellschaft: Hamburg > >> Registergericht: Amtsgericht Hamburg, HRB 120189 > >> Geschäftsführer: Carsten Bock > >> Ust-ID: DE279344284 > >> > >> Hier finden Sie u
Re: [SR-Users] Kamailio Active / Active HA.
On 30 December 2014 at 22:49, Måns Nilsson wrote: > Subject: Re: [SR-Users] Kamailio Active / Active HA. Date: Mon, Dec 29, > 2014 at 11:13:34AM + Quoting Ben Langfeld (b...@langfeld.co.uk): > > > Just a word of caution here, there are several SIP stacks which don't > > handle SRV lookups correctly. I don't mean to discourage you from this, > but > > just to be aware of it. For example, Mobicents only understands SRV > record > > sets which return domain names. If you return IPs, Mobicents will go > ahead > > and look them up as A records and fail. > > I would never, ever, put an IP address in the Target field of a SRV > record. The specification says: > >Target > The domain name of the target host. There MUST be one or more > address records for this name, the name MUST NOT be an alias (in > the sense of RFC 1034 or RFC 2181). Implementors are urged, but > not required, to return the address record(s) in the Additional > Data section. Unless and until permitted by future standards > action, name compression is not to be used for this field. > > RFC2782, p3 > > Pretty clear. A host name there should be, resolved to an /A > record. Mobicents is actually Doing The Right Thing; even though one could > claim the Robustness Principle in favour of the more lenient behaviour, > it is indeed proper to expect to be able to do DNS /A lookups on > the Target string. > Ah, I was not aware of this. Thanks for pointing it out :) > Further, it is a syntax violation to put an IPv6 Address in the Target > field. I made a copy of my zone file and threw in a bogus SRV record. Then > I gave it to the BIND 10 syntax checker, which most predictably barfed > on it: > > ball-empfang:tmp mansaxel$ /usr/local/bind10-0a1/sbin/named-checkzone > besserwisser.org bsu > bsu:35: warning: 2001:470:28:842:201:2eff:fe48:86aa.besserwisser.org: bad > name (check-names) > zone besserwisser.org/IN: _knolk._tcp.besserwisser.org/SRV > '2001:470:28:842:201:2eff:fe48:86aa.besserwisser.org' has no address > records (A or ) > zone besserwisser.org/IN: loaded serial 2014122801 > OK > ball-empfang:tmp mansaxel$ sed -n 35p bsu > _knolk._tcp 10 SRV 0 0 567 2001:470:28:842:201:2eff:fe48:86aa > > So, dont! ;-) > > /Måns, who has a population of specialised IP phone devices that most > certainly won't do SRV. > -- > Måns Nilsson primary/secondary/besserwisser/machina > MN-1334-RIPE +46 705 989668 > LOOK!! Sullen American teens wearing MADRAS shorts and "Flock of > Seagulls" HAIRCUTS! > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Redirect Server Including Path/Recieved Information
For the ease of future reference, it would appear that post was http://sr-dev.sip-router.narkive.com/bfyDpQ36/git-alexh-master-core-modules-tm-modules-sl-make-adding-path-and-flags-to-redirected-contacts#post4 On 9 January 2015 at 09:32, Alex Hermann wrote: > On Thursday 08 January 2015, Asgaroth wrote: > > I am attempting to setup a standalone redirect server which will lookup > > contact info and redirect to appropriate outbound proxy. > > > > The problem I am having is that the registrar is storing the path and > > recieved information, however, when I perform a lookup and reply with > > 302, the recieved and path information is not included. > > > > > Any suggestions/comments to assist in how I get this info into the 302 > > message would be greatly appreciated. > > See my messages from 2011-08-08 on sr-dev mailinglist. It has an > implementation for this. > > [sr-dev] git:alexh/master: core modules/tm modules/sl: Make adding path and > flags to redirected contacts optional > > -- > Greetings, > > Alex Hermann > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] WebRTC to PSTN call, proxied through Kamailio
Maybe you could include you config also? On 10 February 2015 at 15:01, Rahul MathuR wrote: > Hello gents, > > I was trying my hands on getting a successful RTCweb call (JSsip, since > Peter Dunkley mentioned that he's been using JSsip for most of the testing > scenarios..) to PSTN, making my kamailio as proxy + protocol converter (sip > over web-sockets to sip over udp). > And yes, I've referred Carlos' config; the main problem is I get 'Bad > Media Description' error in Google Chromium (Version 40.0.2214.111 m) & > my SIP server even sends 200 OK, but my phone doesn't ring. To make it > worse, I can see rtpengine throwing this error - > "SRTCP output wanted, but no crypto suite was negotiated" > > BTW, I have - > [root@localhost log]# openssl version > OpenSSL 1.0.1j 15 Oct 2014 > > I even tried building kamailio & rtpengine using this openssl but in-vain. > One thing that baffles me is that, apparently kamailio has started > receiving RTP packets (perhaps early media) but the mobile phone hasn't > ringed :-( > > I am attaching all possible logs & seek some guidance from the array of > experts in this list. > > Files attached: > a) tcpdump on ext. interface > b) tcpdump on loopback > c) syslogs > d) Chromium JS logs > > UAC (14.98.55.38), Kamailio (125.99.186.126), SIP Server > (157.238.178.153), Media Server (199.27.244.6) > > > > -- > Warm Regds. > MathuRahul > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] dialog module 4.2 REFER
The REFER's contact header should be the referring party, and is used as the destination for NOTIFYing progress of the refer. The party to refer *to* is stated in the ReferTo header. In what way does the refer fail? Maybe you could provide logs... On 15 February 2015 at 10:38, Uri Shacked wrote: > Hi, > > I am trying to use dlg_refer. I set the side to refer and the final > destination. > But, the contact header of the refer stays "contro...@kamailio.org". So, > the refer fails. > Is it a bug or should i change the contact header by myself before doing > the dlg_refer? > > same with the dlg_bridge. > > > Cheers, > Uri > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio and rtpengine Load Testing
On 12 March 2015 at 11:30, symack wrote: > SIPP is capable of playing media? > Yes it is. Take a look at SippyCup to make it easier: http://mojolingo.github.io/sippy_cup/ > > N. > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Asking for help in configuration Kamailio
Post your request to busin...@lists.kamailio.org On 27 March 2015 at 09:04, Mikael Sarkisyan wrote: > Can i ask you to tell me where i can buy this prof service please > > thans a lot for fast response > > > 27 марта 2015 г., в 14:21, Jon Bonilla (Manwe) > написал(а): > > > > El Fri, 27 Mar 2015 13:21:00 +0300 > > Mikael Sarkisyan escribió: > > > > > >> > >> UAC going to Kamailio IP address and Kamailio forwarding request to > Asterisk > >> IP without auth on Kamailio and auth with asterisk .conf files (i mean > >> without mysql at all) But i wanna keep data about what to do will > presence > >> and other stuff Can you make changes in attached files please? > >> > >> > > > > > > IMHO, you're not asking for help but for consultancy. You should purchase > > professional services for that. > > > > cheers, > > > > Jon > > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Error configuring kamailio with sql.
What steps have you taken so far to address the issue? Do you understand what the error is telling you? On 13 April 2015 at 06:48, Satish Verma wrote: > Respected sir, > I want to configure SIP voip server using tutorial ( > http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour). > i have configured everything on my ubuntu server as given in tutorial. but > i am getting error while running start command ( /etc/init.d/kamailio start) > the error as given below. > > > root@wifi:/etc/kamailio# /etc/init.d/kamailio start > * Not starting Kamailio SIP Server: invalid configuration file! > * > * 0(32729) INFO: tls [tls_init.c:401]: init_tls_compression(): tls: > init_tls: disabling compression... > 0(32729) ERROR: [cfg.y:3288]: yyparse(): cfg. parser: failed to > find command force_rtp_proxy (params 0) > 0(32729) : [cfg.y:3428]: yyerror_at(): parse error in config file > /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing > loadmodule? > > 0(32729) ERROR: [cfg.y:3288]: yyparse(): cfg. parser: failed to > find command force_rtp_proxy (params 0) > 0(32729) : [cfg.y:3428]: yyerror_at(): parse error in config file > /etc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing > loadmodule? > > ERROR: bad config file (2 errors) > > > > i have tried running using default kamailio.cfg then it works fine. but i > want to configure using script uploaded by u. please do the needful. i will > be very thankful to you. > > With regards, > Satish verma, > Programming Assistant, > UIET, Panjab University, > Chandigarh. Mobile no +917307737732 > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Shared location database used by independend kamailios
Note that you could equally use add_path().
On 14 April 2015 at 13:33, Daniel Tryba wrote:
> On Tuesday 14 April 2015 17:11:45 Olle E. Johansson wrote:
> > > I could use sqlops to fetch this manually, but is there an easier way I
> > > am
> > > missing?
> >
> > Use the path header?
>
> When registering directly to the srv records loadbalanced sip servers
> there is
> no path. But forcefully adding it before save() does the trick
>
> if(!is_in_subnet($si, "loadbalancersubnet"))
> {
> append_hf("Path: \r\n");
> msg_apply_changes();
> }
>
> if (save("location"))
>
> (maybe better to check for the absence of a path header)
>
> Next problem is getting to the INVITE to the endpoint. server2 is trying to
> deliver the INVITE to the internal ipadress of uac2.
>
> I hate NAT :(
>
> But thank you Olle for one of your always insightful hints.
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine and security
You might want to read up on ICE (STUN & TURN) and SRTP / DTLS which broadly resolve your issues. On 21 April 2015 at 23:40, GG GG wrote: > By port closed, I mean that ports are normally closed, but when rtpengine > send the first rtp packets to the client, it opens a pinhole in the > firewall, and the matching incoming packets from the client will make the > connection established,related in iptables. I think symmetric nat permits > that. > > But now I'm thinking that it's impossible for rtpengine to know the > client's destination port at the learning phase if the client's rtp packets > can't reach rtpengine. > > Rtpengine can learn the IP Address from kamailio through the --sip-source > CLI switch, but can't guess the port, right ? > > So, playing with established,related is not possible. > > > If the attacker is fast enough, yes. You can disable learning of > > endpoint addresses using the asynchronous flag, but obviously this will > > break NAT'd media. You can also use the strict-source flag to make > > rtpengine drop packets received from a mismatched source address. > > So if I don't use strict-source flag, an attacker could merge any garbage > of data in an existing RTP stream ? > > Thanks. > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405.0 To: From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: Expires: 3600 <-> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405.0;received= kamailio From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: ;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER = sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
Thanks for your response. I did read the section about the secret in the kb url. I followed the example and inserted the test users on tFe url (101, 102, 103) and they have secret set to NULL. I have tried both secret=NULL and secret="" and Asterisk still asks for authentication. Also when I do "sip show peers" I get: Name/username HostDyn Forcerport ACL Port Status Description Realtime kamailio-inbound kamailioIP a 5060 Unmonitored I added qualify=yes and now: Name/username HostDyn Forcerport ACL Port Status Description Realtime kamailio-inbound kamailioIP a 5060 UNREACHABLE Could this be the issue? I have verified that Kamailio receives the responses by doing ngrep and I can see the SIP 401 from Asterisk. Maybe I am missing something else? I'm not sure I understand how Asterisk's peer selection affects this. When I received the registration request from Kamailio, the From: address and domain are the same as the To: address and domain, which are the values I have set in the sipusers table. Another thing, even though the client handset says registered, the table 'sipregs' is not updated with fullcontact, regseconds, or any data at all. Yet I can still make a call. So maybe Asterisk is not authenticating INVITES (whether or not it's registered) and that's why I can call. Any further help or things I should try? Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo < alberto.sagr...@avanzada7.com> wrote: > You could remove secret= on extensiones to check if its related to > authentication or not > > You must not request authentication to kamailio in order to work properly > in front of Asterisk > > As Daniel mention check if Kamailio peer is created and extensiones have > no secret.. you would need to add alternate sippasswd table for kamailio > authentication > > BR > > 2015-07-16 1:42 GMT+02:00 Ben Fitzgerald : > >> Hi, I've been following this integration tutorial >> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb >> and have a successful registration and I can even make calls through my >> asterisk box. >> >> However what is unusual to me is that every time a phone registers with >> Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies >> with 401 Unauthorized. Oddly enough the phone registers and can still make >> calls. What worries me is that as we scale to 100's of cps, this seemingly >> erroneous message may slow down Asterisk because it's trying to handle >> authentication for users which have already been authenticated by Kamailio. >> If this behavior is expected, then that would be good to know as well. >> >> This is the sip debug from ASTERISK (I have replaced IP's with the names >> of the servers): >> >> >> <--- SIP read from TCP:kamailio:41205 ---> >> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 >> Via: SIP/2.0/TCP >> kamailio;branch=z9hG4bK998f.2846e405.0 >> To: >> From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >> CSeq: 10 REGISTER >> Call-ID: 0005ce130bcee5c4-26538@kamailio >> Max-Forwards: 70 >> Content-Length: 0 >> User-Agent: kamailio (4.3.0 (x86_64/linux)) >> Contact: >> Expires: 3600 >> >> <-> >> --- (11 headers 0 lines) --- >> Sending to kamailio:5060 (no NAT) >> Sending to kamailio:5060 (no NAT) >> >> <--- Transmitting (no NAT) to kamailio:5060 ---> >> SIP/2.0 401 Unauthorized >> Via: >> SIP/2.0/TCP >> kamailio;branch=z9hG4bK998f.2846e405.0;received= >> kamailio >> From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >> To: ;tag=as404bac9a >> Call-ID: 0005ce130bcee5c4-26538@ kamailio >> CSeq: 10 REGISTER >> Server: Asterisk PBX 11.6-cert2 >> Allow: INVITE, ACK, CANCEL, OP
Re: [SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
at 11:48 AM, Alberto Sagredo <
alberto.sagr...@avanzada7.com> wrote:
> Maybe you got to get some traces with sip set debug on on asterisk or
> ngrep in kamailio to check whereis the problem.
>
> I think you are not authenticating correctly
>
> Check if you insert on sipusers and sipppers table what is commented on KB
> by asipto.
>
> Maybe your Kamailio is not responding to OPTIONS (qualify=yes)
>
> add at the beginning of your kamailio.cfg file
> request_route {
>
> if(is_method("OPTIONS") ) {
>
> sl_send_reply("200","Keepalive");
>
> exit;
>
> }
>
> .
>
>
> To solve qualify problem
>
>
> BR
>
> 2015-07-16 19:31 GMT+02:00 Ben Fitzgerald :
>
>> Thanks for your response.
>>
>> I did read the section about the secret in the kb url. I followed the
>> example and inserted the test users on tFe url (101, 102, 103) and they
>> have secret set to NULL. I have tried both secret=NULL and secret="" and
>> Asterisk still asks for authentication. Also when I do "sip show peers" I
>> get:
>>
>> Name/username HostDyn
>> Forcerport ACL Port Status Description
>> Realtime
>> kamailio-inbound kamailioIP a
>> 5060 Unmonitored
>>
>> I added qualify=yes and now:
>>
>> Name/username HostDyn
>> Forcerport ACL Port Status Description
>> Realtime
>> kamailio-inbound kamailioIP a
>> 5060 UNREACHABLE
>>
>> Could this be the issue? I have verified that Kamailio receives the
>> responses by doing ngrep and I can see the SIP 401 from Asterisk.
>>
>> Maybe I am missing something else? I'm not sure I understand how
>> Asterisk's peer selection affects this. When I received the registration
>> request from Kamailio, the From: address and domain are the same as the To:
>> address and domain, which are the values I have set in the sipusers table.
>>
>> Another thing, even though the client handset says registered, the table
>> 'sipregs' is not updated with fullcontact, regseconds, or any data at all.
>> Yet I can still make a call. So maybe Asterisk is not authenticating
>> INVITES (whether or not it's registered) and that's why I can call.
>>
>> Any further help or things I should try?
>>
>> Benjamin Fitzgerald
>> LETS Corporation
>> (925) 235-1154
>> b...@letscorp.us
>>
>>
>>
>>
>> ***Confidential Notice:
>> This message is intended only for the use of the individual or entity to
>> which it is addressed and may contain information that is privileged,
>> confidential and exempt from disclosure under applicable law. If the reader
>> of this message is not the intended recipient, you are hereby notified that
>> any dissemination, distribution or copying of this communication is
>> strictly prohibited. If you have received this message in error, please
>> delete this message from all computers and contact Orion Systems/LETS Corp
>> immediately by return e-mail and/or telephone at (925) 566-5600
>>
>> On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo <
>> alberto.sagr...@avanzada7.com> wrote:
>>
>>> You could remove secret= on extensiones to check if its related to
>>> authentication or not
>>>
>>> You must not request authentication to kamailio in order to work
>>> properly in front of Asterisk
>>>
>>> As Daniel mention check if Kamailio peer is created and extensiones have
>>> no secret.. you would need to add alternate sippasswd table for kamailio
>>> authentication
>>>
>>> BR
>>>
>>> 2015-07-16 1:42 GMT+02:00 Ben Fitzgerald :
>>>
>>>> Hi, I've been following this integration tutorial
>>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>>> and have a successful registration and I can even make calls through my
>>>> asterisk box.
>>>>
>>>> However what is unusual to me is that every time a phone registers with
>>>> Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies
>>>> with 401 Unauthorized. Oddly enough the phone registers and can still make
>>>> calls. What worries me is that as we scale to 100's of cps, this seemingly
>>>> erroneous message may
Re: [SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
I think I have fixed the authentication issue yet the SIP dialog has completely changed. Now the dialog involves Asterisk sending SIP NOTIFY to Kamailio, which is then forwarded to the client. Kamailio.cfg has no routes to handle NOTIFY and there are no SUBSCRIBE messages preceding the NOTIFY. Only REGISTER and 200 OK. Is this expected behavior? The sipregs database is now correctly updated when a peer registers so that's good. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Thu, Jul 16, 2015 at 2:59 PM, Ben Fitzgerald wrote: > Thank you for the qualify solution, that worked. > > However, on the KB by asipto, they only create a `sipreg` and `sipusers` > table and then in extconfig.conf for asterisk, sipusers and sippeers are > both using the `sipusers` table in MySQL. > > I included a sip trace in the original email but I will include a more > detailed sip debug here. It looks like Asterisk and Kamailio can exchange > messages but for some reason, the SIP dialog stops after Asterisk sends > back a SIP 401 Unauthorized to Kamailio. Any ideas? > > *1. Kamailio using sipgrep* > > T 2015/07/16 14:50:52.393582 UserAgentIP:64521 -> KamailioIP:5060 > [AP] > REGISTER sip:opvpnx.ulets.us SIP/2.0. > Via: SIP/2.0/TCP 192.168.0.179:64521;alias;branch=z9hG4bK.j~V~btADL;rport. > From: ;tag=QZ7de-7u5. > To: sip:1...@opvpnx.ulets.us. > CSeq: 29 REGISTER. > Call-ID: puXkrkIICT. > Max-Forwards: 70. > Supported: outbound. > Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml. > Contact: > UserAgentIP:64521;transport=tcp>;+sip.instance="". > Expires: 3600. > User-Agent: Alpha TalkIphone/2.2.5-80-g783bf67 (belle-sip/1.4.0). > Content-Length: 0. > Authorization: Digest realm="opvpnx.ulets.us", > nonce="VagoaFWoJzylK0MxoOAIPTRhtZBlmVmr", username="102", uri="sip: > opvpnx.ulets.us", response="24b8f292fca38e72fbcf36417dcecd24". > . > > > T 2015/07/16 14:50:52.440789 KamailioIP:5060 -> UserAgentIP:64521 > [AP] > SIP/2.0 200 OK. > Via: SIP/2.0/TCP 192.168.0.179:64521 > ;alias;branch=z9hG4bK.j~V~btADL;rport=64521;received= UserAgentIP. > From: ;tag=QZ7de-7u5. > To: sip:1...@opvpnx.ulets.us;tag=723cfa83f1495d1e63c1f1bb20bde818.a56d. > CSeq: 29 REGISTER. > Call-ID: puXkrkIICT. > Contact: UserAgentIP:64521;transport=tcp>;expires=3600;received="sip: > UserAgentIP:64521;transport=tcp";+sip.instance="". > LETSSBC. > Content-Length: 0. > . > > *#* > *# These next two messages when Kamailio forwards REGISTER to Asterisk* > *#* > > T 2015/07/16 14:50:52.466461 KamailioIP:43488 -> AsteriskIP:5060 > [AP] > REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0. > Via: > SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24.0. > To: . > From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-3497. > CSeq: 10 REGISTER. > Call-ID: 2ee5ec48557bba33-31464@ KamailioIP. > Max-Forwards: 70. > Content-Length: 0. > User-Agent: kamailio (4.3.0 (x86_64/linux)). > Contact: . > Expires: 3600. > . > > > T 2015/07/16 14:50:52.494578 AsteriskIP:5060 -> KamailioIP:43488 > [AP] > SIP/2.0 401 Unauthorized. > Via: > SIP/2.0/TCP > KamailioIP;branch=z9hG4bK328c.29246e24.0;received= > KamailioIP. > From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-3497. > To: ;tag=as0eb2442e. > Call-ID: 2ee5ec48557bba33-31464@ KamailioIP. > CSeq: 10 REGISTER. > Server: Asterisk PBX 11.6-cert2. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH. > Supported: replaces, timer. > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b30f8aa". > Content-Length: 0. > > *2. Asterisk using sip set debug on* > > t91*CLI> > > <--- SIP read from TCP: KamailioIP:43488 ---> > REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0 > Via: > SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24.0 > To: > From: ;tag=32fda68bf54efeeb04e3edc67b53c63d-3497 > CSeq: 10 REGISTER > Call-ID: 2ee5ec48557bba33-31464@ KamailioIP >
Re: [SR-Users] Check if user is registered before reply is routed
Hi,
Do you know where I can find the example kamailio.cfg file used in that
presentation?
Thanks
Benjamin Fitzgerald
LETS Corporation
(925) 235-1154
b...@letscorp.us
***Confidential Notice:
This message is intended only for the use of the individual or entity to
which it is addressed and may contain information that is privileged,
confidential and exempt from disclosure under applicable law. If the reader
of this message is not the intended recipient, you are hereby notified that
any dissemination, distribution or copying of this communication is
strictly prohibited. If you have received this message in error, please
delete this message from all computers and contact Orion Systems/LETS Corp
immediately by return e-mail and/or telephone at (925) 566-5600
On Thu, Jul 30, 2015 at 3:19 PM, Seudin Kasumovic <
seudin.kasumo...@gmail.com> wrote:
> Helo,
>
> try with tsilo module see excelent presentation here
> https://www.youtube.com/watch?v=4XIrR9bwUkM
>
> Seudin
>
> On Mon, Jul 27, 2015 at 10:39 AM, Dmytro Bogovych <
> dmytro.bogov...@gmail.com> wrote:
>
>> Greetings.
>> Sorry for previous false letter.
>>
>> I want to check if user is registered before reply is routed to its
>> target user agent.
>>
>> Imho it should be like this:
>>
>> onreply_route[MANAGE_REPLY] {
>> if ($rm=="INVITE") {
>> if (registered("location", $ru)) {
>> # Do job - pause for 15 seconds and give chance to reregister
>> }
>> }
>> }
>>
>> But kamailio complains on registered function
>> 0(6274) ERROR: [cfg.y:3295]: yyparse(): misused command registered
>> 0(6274) : [cfg.y:3439]: yyerror_at(): parse error in config
>> file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35:
>> Command cannot be used in the block
>>
>> Documentation confirms - registered() function can be used in
>> REQUEST_ROUTE and FAILURE_ROUTE only.
>>
>> Is it possible to check if user is registered when reply is processed
>> in scripts?
>>
>> Thank you!
>>
>> ___
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>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> --
> MSC Seudin Kasumovic
> Tuzla, Bosnia
>
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>
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[SR-Users] Kamailio/Asterisk Integration - MySQL driver error
Has anyone seen this error before? " ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Host '192.168.xxx.xxx is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts'" I fixed it using flush hosts but I'm curious why this popped up? Is there a way to limit connections for Kamailio and why am I reaching that limit with *only 2* SIP endpoints? This is not a heavily utilized database by any means. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio/Asterisk Integration - MySQL driver error
Thanks for your quick reply. I left it default, my config is like this: fork=yes children=4 and if I check the process list for Kamailio it looks like there are 11 processes. Even so, this is not near the maximum connections supported by MySQL. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Mon, Aug 17, 2015 at 9:54 AM, Alex Balashov wrote: > Ben, > > AFAIK, every Kamailio child process creates its own MySQL connection > handle. What are your 'children' set to? > > -- > Alex Balashov | Principal | Evariste Systems LLC > 303 Perimeter Center North, Suite 300 > Atlanta, GA 30346 > United States > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio/Asterisk Integration - MySQL driver error
Alex, There are 15 total connections and 8 of them are from Kamailio. my.cnf limit is currently set to 500. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Mon, Aug 17, 2015 at 10:01 AM, Alex Balashov wrote: > Ben, > > if you look at the MySQL server, what does netstat tell you, e.g. > > netstat --inet -n | grep 3306? > > -- Alex > > > -- > Alex Balashov | Principal | Evariste Systems LLC > 303 Perimeter Center North, Suite 300 > Atlanta, GA 30346 > United States > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio/Asterisk Integration - MySQL driver error
I should add, I fixed the error and restarted Kamailio - so it's possible that Kamailio had way more connections at the time of the error but I did not check the stats at that time, I was focused on fixing it. However, I would think it's unlikely for Kamailio to create so many connections. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Mon, Aug 17, 2015 at 10:06 AM, Ben Fitzgerald wrote: > Alex, > > There are 15 total connections and 8 of them are from Kamailio. my.cnf > limit is currently set to 500. > > Benjamin Fitzgerald > LETS Corporation > (925) 235-1154 > b...@letscorp.us > > > > > ***Confidential Notice: > This message is intended only for the use of the individual or entity to > which it is addressed and may contain information that is privileged, > confidential and exempt from disclosure under applicable law. If the reader > of this message is not the intended recipient, you are hereby notified that > any dissemination, distribution or copying of this communication is > strictly prohibited. If you have received this message in error, please > delete this message from all computers and contact Orion Systems/LETS Corp > immediately by return e-mail and/or telephone at (925) 566-5600 > > On Mon, Aug 17, 2015 at 10:01 AM, Alex Balashov > wrote: > >> Ben, >> >> if you look at the MySQL server, what does netstat tell you, e.g. >> >> netstat --inet -n | grep 3306? >> >> -- Alex >> >> >> -- >> Alex Balashov | Principal | Evariste Systems LLC >> 303 Perimeter Center North, Suite 300 >> Atlanta, GA 30346 >> United States >> >> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio/Asterisk Integration - MySQL driver error
The language of the error is ambiguous. "many connection errors" could mean lots of connections, or lots of errors related to connections. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Mon, Aug 17, 2015 at 10:08 AM, Alex Balashov wrote: > Oh, I misread your initial post. The error message is about connection > "errors", not excessive connections. > > > -- > Alex Balashov | Principal | Evariste Systems LLC > 303 Perimeter Center North, Suite 300 > Atlanta, GA 30346 > United States > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio/Asterisk Integration - MySQL driver error
Actually I think I've found the issue. The two servers are connected via two different ethernet interfaces. I initially setup the Kamailio user with the IP of eth1 and that worked but Kamailio was trying to connect to MySQL using eth0 (which I did not have that IP authorized to connect). By trying to connect over and over and failing, MySQL blocked the connection ( http://dev.mysql.com/doc/refman/5.0/en/blocked-host.html). Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Mon, Aug 17, 2015 at 10:12 AM, Ben Fitzgerald wrote: > The language of the error is ambiguous. "many connection errors" could > mean lots of connections, or lots of errors related to connections. > > Benjamin Fitzgerald > LETS Corporation > (925) 235-1154 > b...@letscorp.us > > > > > ***Confidential Notice: > This message is intended only for the use of the individual or entity to > which it is addressed and may contain information that is privileged, > confidential and exempt from disclosure under applicable law. If the reader > of this message is not the intended recipient, you are hereby notified that > any dissemination, distribution or copying of this communication is > strictly prohibited. If you have received this message in error, please > delete this message from all computers and contact Orion Systems/LETS Corp > immediately by return e-mail and/or telephone at (925) 566-5600 > > On Mon, Aug 17, 2015 at 10:08 AM, Alex Balashov > wrote: > >> Oh, I misread your initial post. The error message is about connection >> "errors", not excessive connections. >> >> >> -- >> Alex Balashov | Principal | Evariste Systems LLC >> 303 Perimeter Center North, Suite 300 >> Atlanta, GA 30346 >> United States >> >> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Weight balancing with dispatcher module
Hi,
I am having issues getting the dispatcher to work using weighting. I
have been digging through old posts to the list, and tried all that I
have found, but am still having no luck at all.
So, information...
version: kamailio 4.2.4 (amd64/freebsd) be62bd
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_PTHREAD_MUTEX, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE
65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, select, kqueue.
id: be62bd
Dispatcher configuration...
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 3)
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
Dialplan config..
route[DISPATCH] {
if(!ds_select_dst("1", "9"))
{
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
return;
I am loading the values from a database, which are being loaded fine,
and appear to be being formatted correctly aswell..
++---+-+---+--++--+
| id | setid | destination | flags | priority | attrs |
description |
++---+-+---+--++--+
| 1 | 1 | sipx.x.x.106:5060 | 8 |1 | weight=100 |
|
| 2 | 1 | sip:x.x.x.107:5060 | 8 |1 | weight=0 |
|
++---+-+---+--++--+
The IP's are public, so i've masked them... The following is from kamcmd..
{
NRSETS: 1
RECORDS: {
SET: {
ID: 1
TARGETS: {
DEST: {
URI: sip:x.x.x.107:5060
FLAGS: AP
PRIORITY: 1
ATTRS: {
BODY: weight=0
DUID:
MAXLOAD: 0
WEIGHT: 0
}
}
DEST: {
URI: sip:x.x.x106:5060
FLAGS: AP
PRIORITY: 1
ATTRS: {
BODY: weight=100
DUID:
MAXLOAD: 0
WEIGHT: 100
}
}
}
}
}
}
However, when any call is placed, it will always goto the server ending
107 (with weight 0, which means, as far as I understand, it should be
ignored). I've tried with ; after the weights, this made no change either.
When set to round robin, or via priority, it appears to work fine.
Any ideas?
--
Kind Regards,
-
Ben
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Re: [SR-Users] MySQL Connect Errors
I had these errors randomly popup too I'm not sure if you have the same issue but your description sounds similar. My issue was related to my MySQL server having multiple network interfaces with one configured public and one private and these connection errors were a result of how I set up the mysql user. I had it set up as kamailio@'private.ip' but kamailio was actually attempting connection to the public IP (which was not authorized). If you have multiple network interfaces I would suggest checking kamailio.cfg and the user set up to access mysql and verify your IP's/hostnames and maybe even verify on both boxes what interfaces the connections our going out/coming in on. Benjamin Fitzgerald LETS Corporation (925) 235-1154 b...@letscorp.us ***Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Fri, Oct 16, 2015 at 7:11 AM, John Billings wrote: > Both boxes are running Debian 8. > > I’m still looking at the mysql logs, but I haven’t found anything > relevant, yet. > > > On Oct 14, 2015, at 5:00 AM, sr-users-requ...@lists.sip-router.org wrote: > > Hello, > > do you have centos+selinux? Because I heard about a lot of limits with > the combination of the two. > > Have you checked the mysqld syslog messages, anything relevant there? > > Cheers, > Daniel > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Weight balancing with dispatcher module
Hi Daniel,
Many thanks for taking the time to reply.
I have updated to 4.3.1, and am still having the same issue. However, I
have found a work around to the issue (or at least another way of doing
what i needed to do).
I plan to have a cluster of 5 or more media servers (only have 2 at the
moment, as I am still working through the kamailio/asterisk config) and
wanted to be able to force traffic through 1 media server for the sakes
of tracing and troubleshooting. The weighting method appeared to allow
me to do this. However, after reading through the dispatcher module
readme, I found the flags options will allow me to disable targets by
adjusting the flags within the database.
For reasons I don't fully understand, when set to 50/50, 60/40 etc. the
balancing appears to work fine. Its only when you set it the targets to
100/0 things start to go awry, but the only reason to do this was to
send all traffic to one destination. As setting the flag to 4 against
the target disables routing to it, I can leave the weighting alone, and
just adjust the flag values as and when needed.
Kind Regards,
-----
Ben Bliss
On 16/10/2015 3:26 PM, Daniel-Constantin Mierla wrote:
Hello,
can try with latest version in 4.2 series? There was a fix to weight
based balancing quite some time ago, but it might be after 4.2.4 was
released.
Cheers,
Daniel
On 15/10/15 18:11, Ben Bliss wrote:
Hi,
I am having issues getting the dispatcher to work using weighting. I
have been digging through old posts to the list, and tried all that I
have found, but am still having no luck at all.
So, information...
version: kamailio 4.2.4 (amd64/freebsd) be62bd
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_PTHREAD_MUTEX, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024,
BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, select, kqueue.
id: be62bd
Dispatcher configuration...
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 3)
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
Dialplan config..
route[DISPATCH] {
if(!ds_select_dst("1", "9"))
{
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
return;
I am loading the values from a database, which are being loaded fine,
and appear to be being formatted correctly aswell..
++---+-+---+--++--+
| id | setid | destination | flags | priority | attrs
| description |
++---+-+---+--++--+
| 1 | 1 | sipx.x.x.106:5060 | 8 |1 | weight=100 |
|
| 2 | 1 | sip:x.x.x.107:5060 | 8 |1 | weight=0 |
|
++---+-+---+--++--+
The IP's are public, so i've masked them... The following is from
kamcmd..
{
NRSETS: 1
RECORDS: {
SET: {
ID: 1
TARGETS: {
DEST: {
URI: sip:x.x.x.107:5060
FLAGS: AP
PRIORITY: 1
ATTRS: {
BODY: weight=0
DUID:
MAXLOAD: 0
WEIGHT: 0
}
}
DEST: {
URI: sip:x.x.x106:5060
FLAGS: AP
PRIORITY: 1
ATTRS: {
BODY: weight=100
DUID:
MAXLOAD: 0
WEIGHT: 100
Re: [SR-Users] Kamailio Password Encryption
Hi,
The Auth_db module has the ability to read encypted passwords built into
it. I'd advise reading the manual for the module, and that should show
you how to implement it.
Kind Regards,
-----
Ben Bliss
On 04/01/2016 10:13 AM, micho fr wrote:
Hi Guys,
I have a kamailio server 1.3 with clear text Password in Subscriber
table I wish I can encrypt that password by doing some kind of
combinations between username, domain and let kamailio server be
reading the encrypted password
Is there a way to do it?
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
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[SR-Users] loose_route ACK and BYE
Hi, I've fixed a loose_route problem with a broken gateway. Essentially gateway sends ACKs and BYEs with an unroutable contact header (i.e. it sticks the address of the proxy into the header, rather than the destination UA address). This is all on the same network, so there is not NAT. So by caching the contact URI to for replies back to the gateway, I can populate $ru to forward the ACK or BYE back to the correct UA. However, what I've noticed is that for BYEs, I also need to set the $du variable in addition to $ru. But this is required for ACKs. This is not a problem per se, but I'm interesting to know why - could anybody explain? Cheers, Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] cisco spa502g issues
Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@ Contact:: sip:5546@:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018@10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp::5060 Methods:: 6815 AOR:: 5...@sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db@10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip::1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp::5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cisco spa502g issues
Hi Daniel, I shall have a look at that when Im back in the office on monday :) Cheers, Ben. On 17/03/17 18:04, Daniel-Constantin Mierla wrote: Hello, maybe the new phones do STUN and or the ALG breaks somehow the signaling. You should send the ngrep output taken on sip server for such a call in order to be able to analyze what can happen: ngrep -d any -qt -W byline port 5060 Cheers, Daniel On 17/03/2017 06:32, b...@wtf.com.au wrote: Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@ Contact:: sip:5546@:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018@10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp::5060 Methods:: 6815 AOR:: 5...@sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db@10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip::1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp::5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
