sr-users June 2021

sr-users@lists.kamailio.org

48 participants
64 discussions

ENV Values in cfg
by H Yavari 06 Dec '21

06 Dec '21

Hi to all, Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values. Regards,Hossein

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update max_while_loops in config
by Juha Heinanen 08 Oct '21

08 Oct '21

It is possible to update value of max_while_loops dynamically in config file? For example, there could be a need to increase the max in some while loop, but after the loop is done, decrease it again for other loops. There exists an rpc command to do that, but I haven't found how to do it in config file. I tried @cfg_seti.core.max_while_loops = 200; but got syntax error. -- Juha

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manually append_branch to lookup result
by David Villasmil 07 Sep '21

07 Sep '21

Hello guys, I'm trying to dynamically add a branch after doing lookup. The user is found, but in some cases I need to add a branch and do parallel forking. So i'm basically doing: route[LOCATION] { if (!lookup("location")) { .... } if (something) { route(BRANCH_TO_EXTRA); } route(RELAY); } route[BRANCH_TO_EXTRA] { $fs = MY_SOCKET; append_branch("sip:$tU@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port)); return; } For some reason only the branch appended is being used (I have append_branches=1) Ideas? David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [SR-Users] t_drop_replies not working with t_suspend in failure route
by Huber Andreas 03 Sep '21

03 Sep '21

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Kamailio v5.4.6 Released
by Daniel-Constantin Mierla 13 Jul '21

13 Jul '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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SECSIPID Identity Decode
by Daniel W. Graham 06 Jul '21

06 Jul '21

I am unable to base64url decode the json payload in identity header generated by secsipid. (Using python for test) decoded_payload = url64.decode(‘payload’) UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte Header decodes fine this way but not payload. Is this an issue with the payload encoding? Kamailio 5.5 Daniel W. Graham, CTO CMSInter.net LLC DIRECT (989) 400-4230 INTERNET | TELEPHONE | MANAGED IT

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Kamailio BSD license
by stevecarellccie＠yahoo.com 01 Jul '21

01 Jul '21

Hello, I am trying to use Kamailio in our application which would be deployed in certain environment in addition to a cloud infrastructure. As I understand it, parts of the Kamailio code are covered under the BSD license - are you able to share what those are? I am trying to determine if Kamailio can be used in a customer environment or should it be strictly restricted to cloud environments due to the GPL v2 licenses Thanks,Steve

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install stirshaken module by apt at Debian 10
by Yuriy Nasida 30 Jun '21

30 Jun '21

Hi, I know that it's possible to compile manually but what about apt method to install stirshaken module? I am checking the list of modules using: apt search kamailio But... Nothing looks like stirshaken. Thus, only several modules available and for others we should compile manually ? Please advise Thanks

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Websocket In-dialoge SIP routing failed post network loss due to aliases
by Shahid Hussain 30 Jun '21

30 Jun '21

Hi, Websocket module documentation has a code reference to use aliases for SIP routing. However, aliases will not work in the following setup and situation. 1. Kamailio is configured with active and standby node 2. Ping is implemented from webclient and kamailio responds with pong. 3. Two clients ClinetA and ClientB registered themselves to Kamailio. 4. After SIP negotiation (INVITE-200OK) each client learnt about below aliases. Alias of ClientA: alias=172.27.6.98~58336 Alias of ClientB: alias=172.27.6.98~58342 So normally if ClientB wants to send SIP message to ClientA, SIP URI from ClientB looks like below ACK sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 4. Call is in a connected state. Following is the issue. i. Switchover (or network lost or reboot) at Kamailio happened ii. Due to ping pong both the clients detected network loss individually and re-registered themselves. iii. Aliases of both the clients got changed. New Alias of ClientA: alias=172.27.6.98~ 58346 New Alias of ClientB: alias=172.27.6.98~ 58348 iv. But ClientB doesn’t know that ClientA also re-registered and ClientA’s alias got changed and vice-versa v. Because of this ClientB still sends BYE with Initial alias BYE sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 Would like to know what is the recommended solution for this problem using alias or is it a limitation of using alias? Thanks & Regards, Shahid

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Re: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 29 Jun '21

29 Jun '21

Thanks Muhammad, That seems to have done the trick. I am now able to bridge WebRTC to UDP. Regards, Peter Manley M S shaheryarkh at gmail.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Guidance%20building%20a%20WebRTC%20to%20Asterisk%20Intermediate%0A%20Proxy&In-Reply-To=%3CCAFZQphyZE_tGB9%3D5tFQBtXAv5twMh%3D9aCNFAvveGf0-L%3Dy_qwQ%40mail.gmail.com%3E> Sat Jun 26 03:57:27 CEST 2021 * Previous message (by thread): [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy<https://lists.kamailio.org/pipermail/sr-users/2021-June/112799.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112800> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112800> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112800> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112800> ________________________________ The problem seems to that the kamailio is trying to route it over UDP port using WSS transport, which will never work. You are creating a transport bridge (UDP <-> WSS i.e. stateless transport <-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going to be tricky. I strongly discourage it since things get messy for sequential requests. But you can try the following, Here are some PATH module parameters and functions, you will need to setup. 1. You will need to enable "use_received" parameter to ensure connection tracking. 2. It seems you may have enabled "received_format", so you make sure you do NAT detection and use "handle_ruri_alias" for initial requests. 3. You must also enabled "enable_r2", but with caution, since both ends (i.e. UAC and UAS) can be using same kamailio (proxy) socket for in/out. 4. Make sure to use "add_path_received" instead of "add_path". 5. If using OUTBOUND module then make sure it is loaded BEFORE the PATH module. Thank you. -- Muhammad Shahzad Shafi Burraq Technologies Tel: +49 176 99 83 10 85 On Sat, Jun 26, 2021 at 12:47 AM Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm working on building an RTP Proxy which will allow both regular SIP to > SIP transactions and WebRTC to SIP transactions to an Asterisk Server. > > > > SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > AND > > > > WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > The Kamailio/RTPEngine Proxy is stateless (does not use the tm or > registerer modules), and uses the Path module to replace the Route header > with a Path header. > > > > It works grep for SIP/UDP to SIP/UDP transactions. > > > > HOWEVER... It only partially works for WSS to SIP/UDP transactions. The > REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. > > > > When the Asterisk server sends a SIP Options message, I get the following > error: > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < > sip:l4umfumh at 192.0.2.97<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>;transport=ws;alias=209.141.117.212~59751~6 -> > <null> (IP:172.16.0.48:sp) > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for > WebSocket could not be found > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not > create Via header > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/forward.c:550]: forward_request(): building failed > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/receive.c:425]: receive_msg(): error while trying script > > > > It appears that the WebSocket corresponding to the WSS leg cannot be > located when the request is initiated from the Asterisk Server. > > > > Any thoughts on how I can get this to work in the other direction > (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and > Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite > do what I want it to do (i.e. it appears to act as a registrar server and > doesn't forward REGISTER requests on to another registrar). > > > > > > Thanks and regards, > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

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sr-users June 2021