[Kamailio-Users] Pass calls from/to another realm to gateway

Daniel-Constantin Mierla miconda at gmail.com
Thu Oct 29 12:26:37 CET 2009


Hello,

have you watched the sip traffic with ngrep or wireshark? Basically 
there is not a specific constraint related to local domain or foreign 
domain inside kamailio, routing is simply only matter of your config.

Also, in case kamailio is listening on many IP interfaces, you have to 
monitor all of them to see if the message is sent. If there is 
retransmission means that the request was forwarded, check on your sip 
platform to see if any sip traffic comes (also with a network sniffer) 
and be sure you don't have firewalls in between.

Cheers,
Daniel

On 29.10.2009 7:45 Uhr, Alexander wrote:
>   I've encountered one problem I can not solve :( The situation is 
> following: we've got Kamailio working together with our own SIP 
> platform. Our platform is about various call processing business logic 
> and billing. All calls from SIP users should pass through our 
> platform. For now we successfully can make and receive calls inside 
> our domain - NAT is handled fine in most cases, instant messages are 
> handled and so on. It looks like that:
> - Kamailio receives a call from caller
> - Call is redirected to our platform
> - Platform redirects call back to Kamailio and it looks for callee.
>
>   I've tried to implement that logic. It seems to work in some cases, 
> but I encounter one problem. Sometimes Kamailio can not forward a call 
> to the platform via rewritehostport(). It tries to forward SIP 
> request, but nothing happens - only retransmission handler is called. 
> I mean, after call to t_relay(), I see this:
>
> Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: 
> retransmission_handler : request resending (t=0xb618de58, INVITE 
> sip:200213 at 62.117.120.101:5061 
> <http://sip:200213@62.117.120.101:5061/> SIP/2.0...
> Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: 
> retransmission_handler : request resending (t=0xb618de58, INVITE 
> sip:200213 at 62.117.120.101:5061 
> <http://sip:200213@62.117.120.101:5061/> SIP/2.0...
> Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: 
> retransmission_handler : request resending (t=0xb618de58, INVITE 
> sip:200213 at 62.117.120.101:5061 
> <http://sip:200213@62.117.120.101:5061/> SIP/2.0...
>
> SIP requests seem to be correct, and configuration file works ok from 
> time to time.
>
> The problem I've described happens when Kamailio receives a call from 
> another domain (for example, from sipbroker). Inside our domain 
> everything works fine and INVITE passes to our platform correctly.
>
>   I've attached configuration file and debug output. Files were 
> written for and by Opensips, but with Kamailio I encounter the same 
> problem.
>
>   Is this a problem with configuration file or something else?
> ------------------------------------------------------------------------
>
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-- 
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
* http://www.asipto.com/index.php/sip-router-masterclass/





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