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@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@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@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/