29 Oct
2009
29 Oct
'09
8:45 a.m.
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?
29 Oct
29 Oct
1:26 p.m.
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(a)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/
3:32 p.m.
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(a)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(a)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/
5503
days inactive
5503
days old
2 comments
2 participants
participants (2)
-
Alexander -
Daniel-Constantin Mierla