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List:       asterisk-dev
Subject:    [asterisk-dev] Parsing SIP INFO Method in asterisk
From:       Husnain Taseer <husnain.taseer () gmail ! com>
Date:       2016-02-22 18:44:24
Message-ID: CALkA4hKkungoiQ4YPC6kqVSuQnhmF9G8qYqd+V3Ct0Sv6=F52w () mail ! gmail ! com
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[Attachment #2 (multipart/alternative)]


Hello Folks,
I am using asterisk (mixmonitor) as a recording server for Huawei U1911
Unified Gateway, There is an option available in Huawei Unified gateway to
integrate a third party recording machine by doing some configuration. In
short when call between A and B party via Huawei UG got establish it sends
an INVITE to a pre-configured extension and trunk (in my case
5555@myasteriskserver.com) and starts sending both parties RTP to that
recording server.

When I receive INVITE on asterisk server I only get the caller number.
Information of callee is not there as in this case callee will always be
5555. The good thing about Huawei UG is that it sends SIP INFO packet after
the call established on recording machine (asterisk server). This SIP INFO
has content type text/xml in message body and contains information of both
parties.

Now I want to parse that SIP INFO packet to get and update the information
of callee in the CDR. Is there any way to do it in asterisk ?

I have few methods in my mind to do it like write a script which will sniff
INFO packet, parse it and will update the CDR database. The other thing I
can do is that I can use opensips on the edge and can parse packet there
and then relay the call to the asterisk server for recording. But I want to
do it in asterisk in the dialplan or in source code.

Please advice.

Regards,
*Husnain Taseer*
*VoIP Developer*

[Attachment #5 (text/html)]

<div dir="ltr">Hello Folks,<div>I am using asterisk (mixmonitor) as a recording \
server for Huawei U1911 Unified Gateway, There is an option available in Huawei \
Unified gateway to integrate a third party recording machine by doing some \
configuration. In short when call between A and B party via Huawei UG got establish \
it sends an INVITE to a pre-configured extension and trunk (in my case <a \
href="mailto:5555@myasteriskserver.com">5555@myasteriskserver.com</a>) and starts \
sending both parties RTP to that recording server.</div><div><br></div><div>When I \
receive INVITE on asterisk server I only get the caller number. Information of callee \
is not there as in this case callee will always be 5555. The good thing about Huawei \
UG is that it sends SIP INFO packet after the call established on recording machine \
(asterisk server). This SIP INFO has content type text/xml in message body and \
contains information of both parties.  </div><div><br></div><div>Now I want to parse \
that SIP INFO packet to get and update the information of callee in the CDR. Is there \
any way to do it in asterisk ?</div><div><br></div><div>I have few methods in my mind \
to do it like write a script which will sniff INFO packet, parse it and will update \
the CDR database. The other thing I can do is that I can use opensips on the edge and \
can parse packet there and then relay the call to the asterisk server for recording. \
But I want to do it in asterisk in the dialplan or in source code.<br><br>Please \
advice.<br><br>Regards,</div><div><b>Husnain Taseer</b></div><div><b>VoIP \
Developer</b></div><div><br></div></div>



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