sr-users March 2014

sr-users@lists.kamailio.org

101 participants
166 discussions

Kamailio ICSCF strange forking
by DURECU, OLIVIER (OLIVIER) 31 Mar '14

31 Mar '14

Hi all, I have deployed an IMS based on Kamailio 4.1.2 and I don't manage establishing a call. Users registration is OK but when I initiate the call from A(a)ims.net to B(a)ims.net, I notice a strange I-CSCF behavior The INVITE message goes OK from P-CSCF to S-CSCF then to I-CSCF. Till this point everything is fine... Then the I-CSCF performs its LIR diameter request and gets an Ok response from the HSS with a S-CSCF URI The I-CSCF forwards the INVITE at the received S-CSCF. BUT at the same time the I-CSCF generates a DNS request to resolve ims.net SIP SRV (it gets itself as answer...) and forwards a second INVITE (exact copy from the one sent to the S-CSCF) to itself So I have 2 copies of the same INVITE dialog travelling in the network and messing everything. Any idea of the cause of this behavior . Thanks for your support... Olivier

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Kamailio and Asterisks work on HP-UX ?
by Rizwan Khan 31 Mar '14

31 Mar '14

Can they be installed on an HP UX platform. Considering they are written in C, it shouldnt be a porblem, but has anyone tried that? Thanks, Rizwan Khan

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SIP - WebRTC gateway
by Zappasodi Daniele 31 Mar '14

31 Mar '14

Hello, I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp. Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)? Some months ago I found the architecture described here http://www.kamailio.org/wiki/devel/rtcweb_breaker that proposes to introduce a new RTCWeb Breaker. Is it just a proposal or is Kamailio moving following this approach? If Kamailio really requires a RTCWeb Breaker, what are the main issues against using Doubango webrtc2sip with Kamailio? Performance? Interoperability? License? ... Thanks Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

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How to turn on debugging in kamailio?
by mark li 31 Mar '14

31 Mar '14

is there a way to add debug statements in kamailio.cfg? I'd like to be able to dump some of the variables and also see which path the calls are taking. thanks

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ERROR reply code Failure route
by pars3c 31 Mar '14

31 Mar '14

Hello, i receive many error on the failure route like this " t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 500, received: 487" . Do you know why ? My version of Kamailio is the 4.1.2 and the code is this: failure_route[MANAGE_FAILURE] { if (t_is_canceled()) { exit; } revert_uri(); if (!cr_next_domain("$avp(s:carrier)", "$avp(s:domain)", "$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) { exit; } if (!cr_route("$avp(s:carrier)", "$avp(s:domain)", "$rU", "$rU", "call_id")) { exit; } $avp(s:host)= $rd; t_on_failure("MANAGE_FAILURE"); t_on_branch("MANAGE_BRANCH"); append_branch(); if (!t_relay()) { exit; } } Thanks to all

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Forward copy of Register information to FreeSWITCH
by Alexandr Usov 31 Mar '14

31 Mar '14

I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side. I can't (don't know - how to) use in same way integration with FreeSWITCH. Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration. Maybe I can register user on Kamailio and send additional registration request to FS with src ip changed to kamailio (lan ip)? P.S. Reading FreeSWITCH 1.2 book in progress...

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Patch to support symmetric UDP socket in mi_datagram
by Muhammad Shahzad 31 Mar '14

31 Mar '14

Hi, After wasting most of the day trying to make mi_datagram over UDP socket work. I eventually realize that it does asymmetric UDP communication, which creates a lot of trouble for writing a useful MI script using PERL or Python etc. Anyhow, i go through the module code and was able to write a patch for symmetric UDP communication using mi_datagram. I have tested the patch using both Kamailio 4.1 and trunk branches and it seems to compile and run flawlessly. I am attaching here with this patch for Kamailio community. It would be great if Kamailio developers would review and commit this patch to source trunk. Thank you.

2 5

dialog expire, acc
by Kelvin Chua 28 Mar '14

28 Mar '14

I have a situation here, when a dialog expires, it sends a BYE to both call legs and this does not generate an entry on acc table. however, when i use acc_db_request() on event_route[tm:local-request], i get 2 entries on acc, 1 for the downstream, and 1 for the upstream. is there a way of handling this so i can only get 1 BYE on acc? Kelvin Chua

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Kamailio custom header stored in acc db_extra
by Ryan Brindley 28 Mar '14

28 Mar '14

Hey community, I'm trying to get a 2nd leg custom header value stored in my acc table. Right now I'm appending the header with append_hf in a failure route and trying to use: modparam("acc","db_extra","custom=$hdr(Custom)") A sip trace shows the header was properly added and xlog output shows the proper $hdr(Custom) value just after I append the header, but the value doesn't show in acc. Does acc only read from the INVITE of the incoming leg? Any suggestions on how to get this 2nd leg INVITE header in acc is greatly appreciated. Thanks in advance, Ryan

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Trying to dial 4000 into freeswitch's vmail system - Kam Proxy returning Status 403: Not allowed
by mark li 28 Mar '14

28 Mar '14

Hi there. I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as a proxy and registrar ... and freeswitch provides conference calls and voicemail. I have calls between two polycoms working and conference calls work. But when I try to leave a voice message for a user by dialing 44+ext, the sip proxy never replies to the polycom. I did a tcpdump and i can see the INVITE from the polycom to the sip proxy multiple times... but no response back. The phone eventually disconnects itself. Here's what my config looks like: http://pastebin.com/wWgyVcxc Just do a search for "route[FSDISPATCH]". You will see how I check for the "44" prefix, and then send the call to a route called "FSVBOX". Any suggestions would be appreciated. 1. route[FSVBOX] { 2. if(!($rU=~"^1[0-9][0-9]+$")) 3. return; 4. prefix("vb-"); 5. route(FSRELAY); 6. } 7. 8. # Send to FreeSWITCH 9. route[FSRELAY] { 10. $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" 11. + $sel(cfg_get.freeswitch.bindport); 12. route(RELAY); 13. exit; 14. } 15.

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sr-users March 2014