[SR-Users] Problem with parallel forking

Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking. I need to implement parallel forking to different users registered on kamailio. So, when call arrives with R-URI= sip:003912345678@, I need to fork te call to (for example) 2 users: 1001 and 1001. To do that, I've creat

Re: [SR-Users] Problem with parallel forking

mes back due to such loop, using if(src_ip==myself) conditions. Cheers, Daniel On 1/21/11 11:10 AM, Daniel Grotti wrote: Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking.

[SR-Users] Use different certificate for different client with TLS

Hi all, I would like to use kamailio 3.1 with TLS and verified also a client certificate. My tls.cfg file is as follow: --- tls.cfg . . [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key.pem certificate = default_cert.pe

Re: [SR-Users] Kamailio - RTPProxy - Asymmetric RTP

Hi Girard, looks like the Provider are sending RTP from port 5392 instead of 5394 as in SDP, rtpproxy send traffic to 5392 as well. Daniel On 10/31/2013 04:13 PM, Lucas Girard wrote: > Hi all, > I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is used. > > I have the syste

Re: [SR-Users] Kamailio - RTPProxy - Asymmetric RTP

which ports should be used. > > > On Thu, 2013-10-31 at 16:18 +0100, Daniel Grotti wrote: > > > Hi Girard, > > > > looks like the Provider are sending RTP from port 5392 instead of 5394 > > as in SDP, rtpproxy send traffic to 5392 as well. > > > >

Re: [SR-Users] Kamailio - RTPProxy - Asymmetric RTP

which changed the RTP port of the peer. Daniel On Thursday, October 31, 2013 22:10 CET, Lucas Girard wrote: > Yes, I have used the r flag, in rtpproxy_manage(). I have not offer or > answer functions, just manage. > > On Thu, 2013-10-31 at 21:41 +0100, Daniel Grotti wrote:

Re: [SR-Users] Capturing-200 Okey(with SDP) from cfg

Hi, what do you mean with "capture from cfg file" ? Do you want to log 200OK message in kamailio log file ? Daniel On Friday, November 1, 2013 08:36 CET, "Surendra" wrote: > Hi all, > > > > I have a requirement , i.e. I need to capture 200 ok(with > SDP) from cfg file

Re: [SR-Users] Kamailio, Keepalived: Kamailio doesn't respond on virtual IP but real IP

Hi, also you can choose the outbound socket to use by setting '$fs' pseudovariable or use "force_send_socket()" function. http://www.kamailio.org/wiki/cookbooks/4.0.x/core#script_operations http://www.kamailio.org/wiki/cookbooks/4.0.x/core#force_send_socket Daniel On 11/12/2013 08:57 AM, Fran

Re: [SR-Users] extract part of string / INVITE

Hi, have a look at Kamailio transformation: http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#uri_transformations Daniel On 11/13/2013 12:19 PM, Daniel-Constantin Mierla wrote: > Hello, > > try: > > $(ru{uri.param,user}{s.substr,3,0}) > > Cheers, > Daniel > > On 11/13/13 12:13 PM

Re: [SR-Users] forward audio to another sip server

Hi Joli, if you need to use an external media-relay, you need to change the IP in the SDP for INVITE (main route) and 200 OK (onreply route): You can try to use the fix_nated_sdp("2", "EXT_IP_ADDR"). http://kamailio.org/docs/modules/4.0.x/modules/nathelper.html#idp15360968 make sure that "EXT_

Re: [SR-Users] Disable NAT for given IP range

Hi, also you can try to use the IPOPS module, for example: if (is_in_subnet("10.0.123.123", "10.0.123.1/24")) { xlog("L_INFO", "it's in the subnet\n"); xlog("L_INFO", "Skip NAT test\n"); ... ... } http://kamailio.org/docs/modules/4.0.x/modules/ipops.html#idp42416 Daniel On 11/13/2

Re: [SR-Users] (no subject)

Rori, this is not the way. Please subscribe to the mailinglist here: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Daniel On Thursday, November 14, 2013 22:11 CET, "Roni |" wrote: > may i join ___ SIP Express Router (SE

Re: [SR-Users] Kamailio returns "405 Method Not Allowed" for SIP MESSAGE request

Hi, are you handling method MESSAGE in your kamailio.cfg ? if(is_method("MESSAGE")) { do something } Daniel On 11/22/2013 03:10 PM, Karthikeyan R wrote: > Is Kamailio supports SIP MESSAGE request? Please let me know how to > configure. > > > > Regards, > > Karthikeyan R > > > > D

Re: [SR-Users] Log files

Hi, you can check the User-Agent reference $ua, if it is equal to "friendly-scanner", just send back a reply with sl_send_reply("200", "OK") Daniel On 11/26/2013 10:53 PM, Joli Martinez wrote: > How can I do this? Is there an article I can reference or something? I am > new to kamailio and n

Re: [SR-Users] Log files

t; > > # main request routing logic > > route{ > > if ($ua=="friendly-scanner") { > sl_send_reply("200","OK"); > exit; > } > > On Nov 26, 2013, at 5:29 PM, Daniel Grotti <mailto:dgr

Re: [SR-Users] Configuration Issue on Kamailio.

Ciao Laura, would be interesting to see the INVITE from kamailo2-->Cisco and see the headers there, as well as the 180/200 from Cisco->kamailio2. As Carsten said, probably Cisco is messing up From/To headers. The 9990 color is not present in any of the INVITEs you provided, so would be nice to

Re: [SR-Users] Configuration Issue on Kamailio.

pplication/sdp. Content-Length: 249. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. s=SIP Call. c=IN IP4 x.x.x.x. t=0 0. m=audio 18838 RTP/AVP 3 101. c=IN IP4 83.147.65.249. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:10. Il 10/08/16 13:33,

Re: [SR-Users] Stress Testing

Hi, a couple of suggestions: you may running out of RTP ports. kamailio may start to block requests if you are using pike module. UAC may start to de-register after a short expires time, so they won't be reachable anymore if they do not refresh their registration. You may check your natping inte

Re: [SR-Users] Stress Testing

] Stress Testing Hi Dan, We have checked all of that and we also not using pike its really strange From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 16 August 2016 17:24 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users

Re: [SR-Users] Change debug level without restarting Kamailio

Hi this shoud work: Fetch the current debug level: # kamctl fifo debug Set level to 2: # kamctl fifo debug 2 Cheers, Daniel n 08/24/2016 04:54 PM, Federico San Martín wrote: Hi guys, sorry for the newbie question, I searched for it and couldn't find any reference, is there a way to chang

Re: [SR-Users] DSCP options

Hi, you can set TOS value: http://www.kamailio.org/wiki/cookbooks/4.4.x/core#tos just map your DSCP value into the TOS value. -- Daniel On 09/08/2016 10:32 PM, Marcelo Bonin wrote: I need to set DSCP values in Kamailio, I couldn't found a way to set in the configuration. Any help will be ap

Re: [SR-Users] auth result -3

Hi Jon, isn't it "invalid user" ? > auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@' http://www.kamailio.org/docs/modules/4.4.x/modules/auth_db.html Daniel On 12/19/2016 03:23 PM, Jon Bonilla (Manwe) wrote: Hi all I'm getting an auth challenge result of -3 and I don'

Re: [SR-Users] auth result -3

Yeah,It was auth_db, not auth.Good night! :PDaniel G.___ 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 behind NAT, ACK to private IP not advertised public IP.

Hi, not sure if I understood it right but, have you defined the advertised_address ? That should be used in Via and RR as well: https://www.kamailio.org/wiki/cookbooks/4.4.x/core#advertised_address Daniel On 12/29/2016 12:09 AM, Pranathi Venkatayogi wrote: I implemented full NAT logic as p

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Hi, have you configured kamailio in order to log to /var/log/kamailio instead of syslog ? https://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration -- Daniel Grotti On 01/02/2017 03:36 PM, Manoj Gupta wrote: Request to all – Please help we are BADLY stuck in this

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

is a standard log line\n"); After restarting the processes, what do you see in /var/log/kamailio ? What do you get from kamctl fifo debug ? -- Daniel Grotti On 01/02/2017 04:13 PM, Manoj Gupta wrote: Hi Daniel, Thanks for the quick revert! Yes we have configured the same in /etc/rsyslo

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

You should add "ims.airtel.in" as kamailio local domain, in your kamailio.domain table. -- Daniel Grotti On 01/02/2017 05:31 PM, Manoj Gupta wrote: HI Daniel, Here is another problem. Kamailio responds with SIP/2.0 478 Unresolvable destination (478/SL) to REGISTER reques

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Kamailio-asterisk doc: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb There are tones of documentation about kamailio out there. Consider to check SIP protocol as well (rfc3261). -- Daniel Grotti On 01/02/2017 06:18 PM, Manoj Gupta wrote: Now we are getting

Re: [SR-Users] Help Asterisk with Kamailio unable to register with remote VOIP providers

Hi, just remove: #!define WITH_ASTERISK From your kamailio.cfg and restart it. -- Daniel Grotti On 01/02/2017 06:36 PM, Manoj Gupta wrote: Hi Daniel, If it is not too much asking, please Can you help me in giving instructions on how to disable kamailio from my asterisk. I am much

Re: [SR-Users] RTPProxy/Mediaproxy issue

Hi, it looks like your platform/network is introducing jitter in RTP packets. Mediaproxy/rtpproxy usual introduce a very low jitter but it has no impact to performances at all. It's hard to say, you should investigate in your platform first, and then in your network devices. For example, when are

Re: [SR-Users] RTPProxy/Mediaproxy issue

Hi Ravi, yes it means that when RTP traffic passes through your media-relay you have traffic, if you don't use media-realy RTP traffic is end-to-end between clients. To check jitter and other values you can capture your SIP/RTP traffic on your kamailio server with "tcpdump" for example and analy

Re: [SR-Users] RTPProxy/Mediaproxy issue

Hi, Yes packet loss is very high. Please investigate around that. Daniel Ravi wrote: >___ >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

Re: [SR-Users] RTPProxy/Mediaproxy issue

Hi Ravi, if you media/rtp proxy is receiving such packet loss, it means that something behind him is cutting the traffic off, somehow. So you should investigate there, in their configuration. If your analysis is right, it seems your loosing around 80% of your packets. Do you have any router, firew

Re: [SR-Users] RTPProxy/Mediaproxy issue

Hi, When you don't use RTPproxy, RTP traffic is sent end-to-end. What is the RTP path in case of client-to-client RTP ? Is it the same ? Daniel On 02/25/2014 06:34 PM, Ravi wrote: > Dear Daniel, > > Thank you for the response, > >> Do you have any router, firewall something between clients

Re: [SR-Users] cnxcc

Rome! Great choice! :) Daniel On 04/08/2014 10:31 AM, Carlos Ruiz Díaz wrote: > Hi Carsten, > > that's right, I'm replying to this while I wait for a tour to the > Vatican City :-D ;-) > > Cheers, > > > On Tue, Apr 8, 2014 at 3:27 AM, Carsten Bock > wrote: > >

Re: [SR-Users] SDP c= IP doesn't get rewritten

Hi, why you don't user functions from rptproxy module ? http://kamailio.org/docs/modules/4.1.x/modules/rtpproxy.html#idp79896 Daniel On 04/11/2014 03:02 PM, Daniel Ciprus wrote: > If you're going to recommend to upgrade/or use something else - this is > not feasible for some weird reasons. Is

Re: [SR-Users] Only a redirect server

On 06/24/2014 12:18 PM, Jack Smith wrote: > Dear friends, > > Just new to this mailing list, new to Kamailio, and new to running a > sip server. > > I would like to try only a redirect server. I had a look at > http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg > which was

Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls

Hi, a call trace could help you to understand why your server is not receiving response (which one actually ?) form the Cisco. Is that because the Cisco didn't receive your SIP message? Or is it because Cisco replied but the response didn't reach your server ? Try to make a sip trace in order to u

Re: [SR-Users] What is q-value ??

Hi, q value is used in location table for serial forking. If you have a subscriber with 2 contact registered you can use the q value to load one contact first (the one with the higher q) and afterwards load and call the second one. Highest is the value stored, highest is the priority. Daniel O

Re: [SR-Users] Kamailio relays ACK of 200 OK to itself

flow ? -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/16/2015 07:23 PM, Dmitry Sytchev wrote: > Hi all! > > I can't make my Kamailio to correctly relay stateful ACKs for 200 > OK. It takes URI from Route:

Re: [SR-Users] Kamailio relays ACK of 200 OK to itself

Hi, please also check if you have set aliases (kamailio.cfg global parameters) to the IP address in the R-URI. -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/16/2015 07:23 PM, Dmitry Sytchev wrote: > Hi all! > >

Re: [SR-Users] MOH SIP SERVERS

Hi, have you read this ? http://www.kamailio.org/docs/modules/4.2.x/modules/mohqueue.html Otherwise you would need a B2BUA (like Asterisk, Freeswitch, Sems) in order to provide moh functionality. -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria

Re: [SR-Users] Expect the kamailio's sip account is unregistered state when the client app is shutdown

Hi, here you are: https://tools.ietf.org/html/rfc3665#section-2.4 -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/20/2015 10:10 AM, Mickael Marrache wrote: > Hi, > > > > You need to configure your client

Re: [SR-Users] Expect the kamailio's sip account is unregistered state when the client app is shutdown

Hi, if the app crashes, it will just de-register the user after the expiration time, since no re-register came in. -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/20/2015 10:34 AM, Filip Malenka wrote: > Thank you

Re: [SR-Users] Expect the kamailio's sip account is unregistered state when the client app is shutdown

Hi, you may want to use: http://www.kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.p.keepalive_timeout -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/20/2015 10:34 AM, Filip Malenka wrote: > Thank you

Re: [SR-Users] Private IP in SDP packet headers

Hi, Did you forget to fix contact and sdp in your onreply_route? Daniel On Jun 5, 2015 3:30 AM, Alex wrote: > > Hello! > > Please help to fix problem with sdp headers > > UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2) > > When i call from UAC to 9002 i received INVIT