Re: [SR-Users] Kamailio Crash

to install for 4.0.6. Only 4.0.7 is in the repo. Anyway, I still believe the trouble is with writing to the DB. Marc On Tue, Jun 16, 2015 at 12:05 PM, Alex Balashov wrote: > On 06/16/2015 12:04 PM, Marc Soda wrote: > > Is it normal for Kamailio to segfault >> > > No. :-)

[SR-Users] Kamailio Crash

Galara cluster for the usrloc DB. This is Kamailio 4.0.6. Thanks, Marc ___ 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 DTLS

I wound up upgrading rtpengine and that resolved this issue. I ran into something new, but I opened an issue for it (here: https://github.com/sipwise/rtpengine/issues/92 if anyone is interested). Sorry for the noise. On Wed, Mar 25, 2015 at 1:46 PM, Marc Soda wrote: > I'm runni

[SR-Users] rtpengine DTLS

roperly: https://gist.github.com/marcantonio/ea0077a5884e4e4b6b45 I see "SRTCP output wanted, but no crypto suite was negotiated" and I've never seen that one before (the SRTCP part). I have this exact config working elsewhere, but this new setup is not worki

[SR-Users] Kamailio and rtpengine Load Testing

I'm wondering if anyone has any experience load testing with Kamailio and rtpengine using WebRTC. I've had success load testing just Kamailio with sipp, but now I'd like to add the media piece and, if possible, do it via WebRTC. Anyone have any thoughts

[SR-Users] Kamailio and rtpengine

How does Kamailio load balance traffic to rtpengine? Is it load based, round robin, etc? The module makes mention of this but I don't see how it works. Also, it talks about weighting the proxies. How is that accomplished? Thanks, Marc __

Re: [SR-Users] WebRTC to PSTN call, proxied through Kamailio

N number made it mandatory to set >> video port to '0' in 183 and 200. However, JSsip was not happy with that >> and cribbed about codec-formats not being present, ergo "Bad Media >> Description". >> >> Marc, >> Could you please share your conf

Re: [SR-Users] WebRTC to PSTN call, proxied through Kamailio

about the media description error, however, the crypto error is probably not a real issue. Richard explained it here: http://lists.sip-router.org/pipermail/sr-users/2014-December/086271.html I corrected the other issues I was having and that one seemed to resolve itself. Hope that helps, Marc O

Re: [SR-Users] NDB_REDIS password to REDIS remote DB

fusion came from the fact that you do not specify the password in the modparam() connection string, as you do with other DB modules. Maybe calling something like this out in the documentation would help clear that up. On Thu, Jan 29, 2015 at 3:48 PM, Marc Soda wrote: > Ah, so you should be able to d

Re: [SR-Users] NDB_REDIS password to REDIS remote DB

does the authentication after connecting > and can be done from config -- that's on a very quick search, not sure if > something has changed with the hiredis api meanwhile: > > - https://github.com/redis/hiredis/issues/56 > > Cheers, > Daniel > > > On 29/01/15 17:20, Ma

Re: [SR-Users] NDB_REDIS password to REDIS remote DB

The only way it will work right now is to not use a password: modparam("ndb_redis", "server", "name=localredis;addr=localhost;port=6379") I've been wanting to look at contributing support at that, but no time... On Thu, Jan 29, 2015 at 10:16 AM, Yuriy Gorlichenko wrote: > Hello. I try to use ND

Re: [SR-Users] Media trouble with kamailio/rtpengine

ut the issue. It was totally client related. Are there plans for rtpengine support trickle ICE anytime soon? My (limited) understand is that without it I can't support Firefox WebRTC clients. Thanks, Marc ___ SIP Express Router (SER) and Kamaili

[SR-Users] rtpengine stats

Does anyone know how a can get stats from rtpengine? I see the $rtpstat pseudo variable in Kamailio, but from the documentation it looks like that will only give me stats on a particular call. I'm looking for overall stats like concurrent calls, bandwidth, etc... Thanks,

Re: [SR-Users] Media trouble with kamailio/rtpengine

Thanks for the response. You're right, the media stream is making it all the way back to my PC, I just don't hear anything. And yes, my speakers are turned up... I'm not sure what to try next... On Fri, Dec 19, 2014 at 12:31 PM, Richard Fuchs wrote: > > On 12/19/14 1

Re: [SR-Users] Media trouble with kamailio/rtpengine

On Fri, Dec 19, 2014 at 10:47 AM, Marc Soda wrote: > > I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not > getting audio back to my browser. From a packet capture I can see media > from the browser to rtpengine, and then bi-directional RTP back and fo

[SR-Users] Media trouble with kamailio/rtpengine

I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser

Re: [SR-Users] SIP Fragments

That's how I ended up going. It's working now. Thanks. On Thu, Dec 18, 2014 at 4:11 PM, James Cloos wrote: > > >>>>> "MS" == Marc Soda writes: > > MS> I'm having a problem reassembling UDP packets on my Asterisk servers > after > M

Re: [SR-Users] SIP Fragments

Content-Length: 1901. So trimming up the headers isn't going to get me anywhere... I'm not comfortable enough with WebRTC to know what to trim out of the SDP, either. How can I force Kamailio to use TCP for SIP when relaying the call? I haven't found much info on it. Marc

Re: [SR-Users] SIP Fragments

So gzcompress is no good with Asterisk then? Is that meant to be used only with another Kamailio proxy? We're trying to do a WebRTC POC with Kamailio as the proxy. The SIP headers and SDP are huge! I've never seen such big messages. Thanks, Marc On Wed, Dec 17, 2014 at 6:47

[SR-Users] SIP Fragments

esn't seem like a realistic solution. I see that Kamailio can compress the message body, but can Asterisk handle that? How do other people handle this? Thanks in advance, Marc ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing lis

Re: [SR-Users] Contact Header User

Finally! This seems to have done it: xlog("L_NOTICE", "$sel(contact.uri.user)"); Sorry for the noob question. On Mon, Dec 8, 2014 at 11:14 AM, Marc Soda wrote: > Can someone recommend a way to extract the user part of a Contact header > URI? > > Right now I&

[SR-Users] Contact Header User

gives me an error starting Kamailio: "function xlog: parameter 2 is not constant". Note, I can't seem to get any select working, for example, @ruri gives me the same error. Any suggestions? Thanks, Marc ___ SIP Express Router (SER) and K

[SR-Users] Plz help: MSILO on TLS + cert based client auth enabled server

recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc ___ SIP Express Router (SER) and Kamailio

Re: [SR-Users] MSILO - dumped messages get duplicated

Hi Daniel! I applied now the source ip check and it is working! Thanks a lot! regards Marc From: proy...@hotmail.com To: sr-users@lists.sip-router.org Date: Tue, 3 Jun 2014 21:38:32 + Subject: [SR-Users] MSILO - dumped messages get duplicated Hi, I have a Kamailio 3.2 setup up

Re: [SR-Users] MSILO - dumped messages get duplicated

Hi, Can anybody help me with these duplicate messages? I guess the first message should be removed from the db, since it got a reply 202. Despite that, it end up again in the db. Can you help me what I am doing wrong? Many thanks Marc From: proy...@hotmail.com To: sr-users@lists.sip

[SR-Users] Trigger m_dump() with REGISTER

registering for 300 secs. Only a REGISTER with an expiration time of 300 should trigger m_dump(). Can you help me with this? Any help or idea is appreciated. thanks Marc ___ SIP Express Router (SER) and

[SR-Users] SDP and NAT

Hey all, I have a Kamailio box functioning as a proxy. When some UAs send an INVITE the SDP has their private IP (c=IN IP4 10.x.x.x). Kamailio is passing this on without changing it to the proper public IP. What's the best way to rewrite that? I'm using a config based on the default one. _

[SR-Users] MSILO - dumped messages get duplicated

. I need to avoid this duplicates but still make sure the message get delivered sooner or later. Any help is highly appreciated. Here is my routing table: thanks Marc ### Routing Logic # Main SIP request routing logic # - processing of any incoming SIP request starts with this

Re: [SR-Users] rport

fer to is generating a new REGISTER from Kamailio to > Asterisk, putting in it the Contact header with the address of kamailio. So > Asterisk should send the INVITE to kamailio. > > Cheers, > Daniel > > > On 28/03/14 19:37, Marc Soda wrote: > > Basically, I'm tryin

Re: [SR-Users] rport

Fri, Mar 28, 2014 at 2:28 PM, Marc Soda wrote: > I have a Kamailio server forwarding REGISTERs to an Asterisk box, similar > to the way Daniel shows here: > > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > > Can I force Kamailio to append rport=5060

[SR-Users] rport

? I tried add_local_rport() but it only included 'rport' and not 5060. Thanks, Marc ___ 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, maddr, and domain

alue and use it as outbound proxy: > > $du = "sip:" + $(ru{uri.maddr}); > > subst_uri() from textops should help removing the parameter. > > Cheers, > Daniel > > > On 17/03/14 17:42, Marc Soda wrote: > > I'm trying to handle a redirect with get_redi

[SR-Users] Redirect, maddr, and domain

I'm trying to handle a redirect with get_redirects(). It seems that Kamailio is ignoring the maddr param on the contact header. Is there a way to force maddr to be used? The Contact header on the 302 looks like this: ;q=0.5,;q=0.25 The message is then being sent to domain.com, rather than 1.1.

Re: [SR-Users] 401 after a 302

I found t_on_branch_failure() in 4.1. Would that be the way to handle this? On Thu, Mar 13, 2014 at 1:36 PM, Marc Soda wrote: > Can someone tell me how to handle a 401 from a 302 redirect? I am > attempting to register with the uac module. Normally, I set a failure > route for th

[SR-Users] 401 after a 302

e_route")) t_on_failure("MANAGE_FAILURE"); } xlog("L_NOTICE","t_relay()'ing ($rm)\n"); if (!t_relay()) { sl_reply_error(); } exit; } Thanks! Marc ___ 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] Incorrect Contact address

but containing a route header pointing to the Kamailio IP. > Kamailio will loose_route() this request and send it to the backend server > as expected. > > Regards, > > > On Wed, Mar 5, 2014 at 3:53 PM, Marc Soda wrote: > >> Thanks Olle. I am calling record_record() o

Re: [SR-Users] Incorrect Contact address

where, it should be rewritten to 2.2.2.2. On Wed, Mar 5, 2014 at 1:34 PM, Olle E. Johansson wrote: > > On 05 Mar 2014, at 18:30, Marc Soda wrote: > > I have Kamailio setup as a proxy in front of a backend server (Asterisk). > When I make a call through the proxy, the Contact

[SR-Users] Incorrect Contact address

ere a special method to rewrite the Contact header to be Kamailio's IP? Where is a good place in the config to do this? (my config is loosely based on this: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) Thanks! Marc _

[SR-Users] INVITE proxy auth

| | | -ACK> | | | | -ACK> | It works, but, it's terrible... Before I try to make it work differently, what do you all think it should do? Marc ___ SIP Express Rout

Re: [SR-Users] REGISTER failure_route 401 problem

On Mon, Mar 3, 2014 at 12:24 PM, Daniel-Constantin Mierla wrote: > > On 03/03/14 18:08, Marc Soda wrote: > > I resolved the issue, but I not quite sure why is worked. Rather than > sending the REGISTER with t_reply() > > > t_reply() is not sending REGISTER anywhere, it

Re: [SR-Users] REGISTER failure_route 401 problem

ded to setup a reply route as well. However, as you can see above, MANAGE_REPLY isn't set for REGISTERs. Why did this fix the problem? Marc On Mon, Mar 3, 2014 at 11:52 AM, Daniel-Constantin Mierla wrote: > Are you sure you have set t_on_failure() for the respective transaction? &

Re: [SR-Users] REGISTER failure_route 401 problem

So I've found out that NAT has nothing to do with it. The bit about things working when the NAT device is removed was wrong. So my question becomes: Why would Kamailio ignore a 401 rather sending it to a failure route? Thanks in advance, Marc On Mon, Mar 3, 2014 at 9:10 AM, Marc Soda

Re: [SR-Users] REGISTER failure_route 401 problem

I forget to mention, the nat device is in front of the Kamailio servers, not the endpoints. On Fri, Feb 28, 2014 at 6:22 PM, Marc Soda wrote: > I have a Kamailio server setup which is registers to a back end server on > behalf of endpoints. The endpoints can register to Kamailio but Ka

[SR-Users] REGISTER failure_route 401 problem

ow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="fe-c7c5-9o.domain.com", nonce="151e4f60" Content-Length: 0 Thanks, Marc ___ SIP Express Router (SE

[SR-Users] MSILO: remove "Offline message" prefix

unchanged? Please advise! Any help is appreciated! Thanks you, Marc ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-b

[SR-Users] TLS: support TLS v1.2 ?

Hi! I have question concerning TLS module. I would like to enable TLS 1.2 protocol. As I understand the TLS module is compiled with openssl, which as far as i know has TLS 1.2 support since openssl 1.0.1. I am wondering if there is way to enable TLS 1.2 on the TLS module? Many thanks Marc

[SR-Users] Msilo: remove messages from db immediately after successful delivery

either: Remove the message from the db immediately after successful delivery (preferred) and/or be able to see the individual message delivery status. Can you help me, how should I proceed? Many thanks in advance! Marc

[SR-Users] Limit the nr. of transaction retries

route. (I gues it is a tm param?) Thansk you for your help in advance! regards Marc ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip

[SR-Users] msilo - use case: registered but not reachable

"OK"); } exit; } else{ m_store("$ru"); sl_send_reply("200", "OK"); } exit; } but i have still the same behaviour. Can you help me, what do I wrong? Many thnaks Marc. M

Re: [SR-Users] t_replicate() and Contact

Also, I tried setting the Contact header in onsend_route, but it seems that it's overwritten after that. Perhaps I need to modify it on the second Kamailio server? Is it possible to change it before it's sent on the first server? On Wed, Sep 4, 2013 at 8:53 AM, Marc Soda wrote:

[SR-Users] t_replicate() and Contact

act header once it is received on 3.3.3.3 is user@2.2.2.2. I think it should be user@1.1.1.1. Where's the best place to change this before it is sent? -- Marc Soda, Sr. Systems Engineer *CoreDial, LLC* | www.coredial.com 1787 Sentry Parkway West, Building 16, Suite 100, Blue Bell, PA 19422 Of

Re: [SR-Users] Loopback

$du = $null; } } Thanks a lot Daniel! On Thu, Aug 29, 2013 at 10:58 AM, Marc Soda wrote: > On Thu, Aug 29, 2013 at 3:53 AM, Daniel-Constantin Mierla < > mico...@gmail.com> wrote: > > >> what device is at 701? The 200ok recev

Re: [SR-Users] Loopback

ontact in 200ok. That can be used to route the ack, > like: > > handle_ruri_alias(); > $ru = $du; > $du = $null; > There is no NAT in this scenario, although NAT support is enabled as we will have to deal with it. Should I still use h

Re: [SR-Users] Loopback

Thanks, I appreciate it. In this setup the there are 2 endpoints (700 and 701) peered up to an Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20). 700 (172.16.60.28) is calling 701 (172.16.3.65). When 701 answers the OK is sent to the proxy and then to Asterisk. Asterisk is then

[SR-Users] Loopback

I think I found my missing ACKs! Can anyone tell me why they work be being sent to the loopback interface? The destination address is still the external (eth0) IP. -- Marc Soda, Sr. Systems Engineer *CoreDial, LLC* | www.coredial.com 1787 Sentry Parkway West, Building 16, Suite 100, Blue

Re: [SR-Users] What does this mean...

Take a look at where route(frompstn) is being called. It's probably in a 'if' statement, meaning that if the source address is the pstn.gw_ip, then return TRUE. Returning -1 is like saying, FALSE. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-

Re: [SR-Users] What does this mean...

I think I found my missing ACKs! Can anyone tell me why they work be being sent to the loopback interface? The destination address is still the external (eth0) IP. On Mon, Aug 26, 2013 at 3:36 PM, Marc Soda wrote: > It's checking the source of the current message. If the source ad

Re: [SR-Users] What does this mean...

ip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Marc Soda, Sr. Systems Engineer *CoreDial, LLC* | www.coredial.com 1787 Sentry Parkway West, Building 16, Suite 100, Blue Bell, PA 19422 Office: (215) 297-4400 x203 | Fax: (215) 297-4401 | Email: ms...

[SR-Users] Config debugging

has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } Why would that be the last line that is traced? Is Kamailio really not processing past here? -- Marc

[SR-Users] (no subject)

nch_route()) { add_rr_param(";nat=yes"); } } } Why would that be the last line that is traced? Is Kamailio really not processing past here? -- Marc ___ SIP Expre

Re: [SR-Users] Rtpproxy newbie question

Hi Carsten, Sorry about the double posting, I had a technical issue on my side and some mails where sent out twice, also that one. Please accept my apology. Thanks a lot for your advice! Regards Marc -Original Message- From: kaiserbo...@googlemail.com [mailto:kaiserbo

[SR-Users] Rtpproxy newbie question

rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an

[SR-Users] Rtpproxy newbie question

rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an

Re: [SR-Users] Siremis 1.0 login fails (or, does nothing)

I had what I think is the same issue. I eventually found that if I clicked the submit button with my mouse instead of hitting enter after entering the password, it would let me login. Very strange and I never got back to figure out what the underlying issue was. Sent via BlackBerry by AT&T -