17 Sep
2008
17 Sep
'08
7:52 a.m.
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
17 Sep
17 Sep
7:57 a.m.
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
--
Daniel-Constantin Mierla
http://www.asipto.com
8:01 a.m.
2008/9/17 Daniel-Constantin Mierla <miconda(a)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
8:05 a.m.
On 09/17/08 15:01, Jon Farmer wrote:
2008/9/17 Daniel-Constantin Mierla <miconda(a)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
--
Daniel-Constantin Mierla
http://www.asipto.com
8:56 a.m.
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.
10:53 a.m.
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
--
Daniel-Constantin Mierla
http://www.asipto.com
11:13 a.m.
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
5:50 p.m.
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
--
Daniel-Constantin Mierla
http://www.asipto.com
5945
days inactive
5945
days old
7 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Jon Farmer