Hi all
I have to following setup for a incoming call
Asterisk -> OpenSER -> 3rd Party SBC
So the call comes into Asterisk which sends it to OpenSER which forwards it to the 3rd party SBC. I need to use a forward() to send to the SBC as the URI needs to be preserved as it came from Asterisk. This is all working fine, however when the called party answers and the OK is sent for the INVITE it is not sent back to Asterisk.
How do I get this to work?
Regards
Jon
Hello Jon,
On 09/17/08 14:52, Jon Farmer wrote:
Hi all
I have to following setup for a incoming call
Asterisk -> OpenSER -> 3rd Party SBC
So the call comes into Asterisk which sends it to OpenSER which forwards it to the 3rd party SBC. I need to use a forward() to send to the SBC as the URI needs to be preserved as it came from Asterisk. This is all working fine, however when the called party answers and the OK is sent for the INVITE it is not sent back to Asterisk.
How do I get this to work?
can you inspect the INVITE sent to SBC and the received 200 OK and see if the Via headers are good? There should be two: one for Asterisk and one for kamailio/openser. Or you can post the sip meesages here to investigate.
Cheers, Daniel
2008/9/17 Daniel-Constantin Mierla miconda@gmail.com
can you inspect the INVITE sent to SBC and the received 200 OK and see if the Via headers are good? There should be two: one for Asterisk and one for kamailio/openser. Or you can post the sip meesages here to investigate.
Is it ok to post attachments to this list? If not I will post the SIP trace in the email body.
Regards
Jon
On 09/17/08 15:01, Jon Farmer wrote:
2008/9/17 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com>
can you inspect the INVITE sent to SBC and the received 200 OK and see if the Via headers are good? There should be two: one for Asterisk and one for kamailio/openser. Or you can post the sip meesages here to investigate.Is it ok to post attachments to this list? If not I will post the SIP trace in the email body.
both are fine -- there is a limit of 40k for the messages posted to the mailing list, no matter there are attachments or inline content.
Cheers, Daniel
Is it ok to post attachments to this list? If not I will post the SIP trace in the email body.
both are fine -- there is a limit of 40k for the messages posted to the mailing list, no matter there are attachments or inline content.
Hi
Here is the SIP trace. The IP's are as follows
PSTN Gateway 10.0.0.1 Asterisk 10.0.1.1:5061 Openser 10.0.1.1:5060 SBC 11.0.0.1
In this trace I am substituing the SBC with X-Lite as I dont have access to the SBC at the moment. But is should illustrate what is going on hopefully.
Thanks for any help that can be given.
Regards
Jon
Hello,
On 09/17/08 15:56, Jon Farmer wrote:
Is it ok to post attachments to this list? If not I will post the SIP trace in the email body. both are fine -- there is a limit of 40k for the messages posted to the mailing list, no matter there are attachments or inline content.Hi
Here is the SIP trace. The IP's are as follows
PSTN Gateway 10.0.0.1 http://10.0.0.1 Asterisk 10.0.1.1:5061 http://10.0.1.1:5061 Openser 10.0.1.1:5060 http://10.0.1.1:5060 SBC 11.0.0.1 http://11.0.0.1
In this trace I am substituing the SBC with X-Lite as I dont have access to the SBC at the moment. But is should illustrate what is going on hopefully.
the 200 ok is correct and seem to be relayed to asterisk, the missing part is in the capture filter - the communication between asterisk and openser is done via loopback I guess, as I don't see the invite from asterisk to openser neither the 200 ok from openser to asterisk. If you are using ngrep give the parameter "-d any" to get the traffic on all network interfaces.
Cheers, Daniel
the 200 ok is correct and seem to be relayed to asterisk, the missing part is in the capture filter - the communication between asterisk and openser is done via loopback I guess, as I don't see the invite from asterisk to openser neither the 200 ok from openser to asterisk. If you are using ngrep give the parameter "-d any" to get the traffic on all network interfaces.
OK well here is the trace with the "-d any" option. Basically when X-Lite answers the call it hangs up immediately at both ends.
Regards
jon
On 09/17/08 18:13, Jon Farmer wrote:
the 200 ok is correct and seem to be relayed to asterisk, the missing part is in the capture filter - the communication between asterisk and openser is done via loopback I guess, as I don't see the invite from asterisk to openser neither the 200 ok from openser to asterisk. If you are using ngrep give the parameter "-d any" to get the traffic on all network interfaces.OK well here is the trace with the "-d any" option. Basically when X-Lite answers the call it hangs up immediately at both ends.
now all the sip messages are in the trace. Very interesting is that the 200ok has the Contact header address with hostname, not with IP and port, which is very unusual for sip. Is the way x-lite is building the reply?
You should do record-routing/route handling in openser, so all the messages within a dialog pass through openser. I see the ACK is going from asterisk to openser, and then seems to be dropped by openser, perhaps you have some logic to do that in your config.
Cheers, Daniel
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Daniel-Constantin Mierla -
Jon Farmer