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37510 discussions

Kamailio issue
by bazou2 bazou2 13 Nov '17

13 Nov '17

Hello, I have a kamailio version 4.4 with an Asterisk 13 ... and i need to make a call from a sip phone app.(i have a user subscribed on the kamailio to be used on the sip phone app). The weird thing that first time, the call works (I can see the sip messages working fine between the app kamailio and asterisk) but when i try again it doesn't work. SIP messages show that the kamailio and Asterisk keep send trying request between them and the loop of these kind of messages doesn't end.. I need to know if i am missing something in the configuration file of the kamailio or what? Regards Sent from Outlook<http://aka.ms/weboutlook>

2 2

Development open in master branch (to be v5.2.x)
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, git branch 5.1 was just created (to host the release series v5.1.x), therefore new features can now be pushed again in master branch. They will be part of the next future release, likely to be numbered 5.2.x. Any fixes that affect existing code in branches 5.1 or older version have to be backported - push first to master and then cherry pick -- see the contributing guidelines at: - https://www.kamailio.org/wiki/devel/git-commit-guidelines#backporting_commi… Many thanks to all contributors so far! Testing of branch 5.1 and giving feedback for it is very appreciated! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Branch 5.1 created
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, the GIT branch 5.1 has just been created, it will host the release series 5.1.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.1 origin/5.1 Hopefully in two-three weeks time frame the full release of 5.1.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.1. No new features can get in branch 5.1. Enhancements to documentation or helping tools, as well as kemi exports are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

GeoIP2 module deb package
by Joel Serrano 12 Nov '17

12 Nov '17

Hi, I can't seem to find the geoip2 module for debian stretch (installing from packages, kamailio 5.0 branch). Is it not available in the repository? Thanks, Joel.

2 4

Kamilio for IMS and VoLTE Setup Questions
by Jack Folgolburg 11 Nov '17

11 Nov '17

Hello, I saw the Kamailio World 2017 presentation "Kamilio for IMS and VoLTE" (https://www.youtube.com/watch?v=Wdhcdv5JVPw). I would like to install the servers for evaluation and testing. The slides say to use Ubuntu 16.04, but there are no xenial packages at the ng-voice repository. Is it okay to install on Ubuntu 14.04, since there are trusty packages? I have done this and got through all the configuration prompts. After installation, is there any way to confirm that the site is operational? How to bring up the management console (i.e. what is local URL)? In the presentation, it looks like a commercial site at ng-voice is accessed. I understand that there is a REST API. Is this API documented anywhere? Maybe, the most important question is, after installation, what is the fastest way to add two users and test access from IMS clients? Thank you, James

1 0

183 acc records even if early_media equals to 0
by Marco Capetta 09 Nov '17

09 Nov '17

Dear All, I'm facing a strange problem with the call accounting module: even if in my configuration I have the parameter: modparam("acc", "early_media", 0) I can find some ACC records with sip_code 180 or 183. I investigated those cases and this issue seems to happen when an endpoint sends an 183 message immediately followed by a 200 OK. The issue seems started from kamailio version 4.4.1, but I cannot be really sure of this. Have you ever had a similar problem? Do you have any suggestions on how to solve it? Thanks Marco

2 11

t_relay_to_udp - 500 server error occurred (6/SL)
by Donat Zenichev 09 Nov '17

09 Nov '17

Hi all. I have a problem with t_relay_to_udp function. My topology: UAC -> tcp -> kamailio <-> udp <-> asterisk Kamailio processes requests from clients over TCP and relays requests to asterisk over UDP. Kamailio gets INVITE from client, relays it to asterisk, then he sends 100 trying to user client and abruptly sends 500 internal error to client (after 100 trying). But the main interesting thing is, that asterisk answers on INVITE and dialog continues between kamailio and asterisk (without user agent client) - it reaches the destination then, without originator : ) How error on kamailio looks like: Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (6/SL) The same problem is, when I change transport protocol between kamailio and asterisk to TCP, using the t_relay_to_tcp function. When I use t_relay function, it works fine, nothing happens that can drop dialogs between kamailio and clients. Relaying code part: if ($var(to_asterisk)=="1") { xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070 \n"); force_send_socket(udp:10.1.1.1:5070); if (!t_relay_to_udp()) { sl_reply_error(); } } I need to use t_relay_to_udp with force_send_socket, because kamailio changes the source port (source socket) every time he generate new INVITEs to asterisk and this is irrelevant. Dump between kamailio and user agent client: INVITE sip:1111111111example@kam1.domain.com:5070;transport=TCP SIP/2.0 Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z- Max-Forwards: 70 Contact: <sip:debug.device-example@uac.domain.example:5065;transport=TCP> To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="debug.device-example",realm=" kam1.domain.com ",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example@kam1.domain.com:5070 ;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a orithm=MD5 Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 v=0 o=Z 0 0 IN IP4 uac.domain.example s=Z c=IN IP4 uac.domain.example t=0 0 m=audio 8000 RTP/AVP 8 3 110 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: MS Lync Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL) Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070 ;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113 From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: example Content-Length: 0 Thanks in advance for hints. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-800 http://wnet.ua

1 0

Topoh module configuration for SIP proxy and SIP registrar
by Christian Conus 08 Nov '17

08 Nov '17

Hello, I am trying to configure the topoh module for kamailio 5.0.2. I read the documentation but some points are still not clear to me. The topology of my network is as follow: - Two SIP proxies - Two SIP registrars - Only one SIP proxy is running (the other take over automatically in case the first fails) - Both registrars are running (proxy sends requests to both registrars using round robin algorithm) Text mode graph of the topology (hope it helps understand): I tried to enable topoh on SIP proxy only, on both proxy and registrars, with same mask_key everywhere and with different mask_key between proxy and registrars but everytime I could still see parts of the SIP header which was not obfuscated. My questions are: - Is it necessary to enable topoh module on each SIP proxy and SIP registrar ? - The mask_key should be the same on each server or it should be different between proxy and registrars? - the mask address should be the same on each server or it should be different between proxy and registrars ? Thanks in advance for your help, Christian -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 10

evapi_async_relay() failed
by Abdul Basit 07 Nov '17

07 Nov '17

Hi list, I am configuring CGRateS with kamailio 4.4 using deb packages on debian 8. I am facing a issue while processing event_route[dialog:start] for answering call INVITE at evapi_async_relay("{\"event\":\"CGR_CALL_START\", with following error /usr/sbin/kamailio[13887]: ERROR: evapi [evapi_mod.c:288]: w_evapi_async_relay(): failed to suspend request processing However, after discussion with Dan on IRC i replaced function evapi_async_relay() with evapi_relay() resolved the issue. Whats causing this error? Can anyone suggest some way out for keep using evapi_async_relay? -- regards, abdul basit

2 1

Kamailio SIP-I
by Saleem Manasfi 07 Nov '17

07 Nov '17

Dears, Please if you can let us know how does Kamailio reads the SIP-I body (ISUP)? Is there any document or link that may help?

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