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 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 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 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?
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?
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Strange... We have no firewalls inside and tcpdump shows outgoing traffic. As I see, it means that Opensips/Kamailio attempts to forward SIP request, but no "Giving a try" is sent back. We'll try to contact gateway vendor.
2009/10/29 Daniel-Constantin Mierla miconda@gmail.com
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?
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla
- Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
- http://www.asipto.com/index.php/sip-router-masterclass/
-
Alexander -
Daniel-Constantin Mierla