[asterisk-users] Capture sip Response

[Sammy Govind]

HI,

*Asterisk 1.8 *allows to read SIP response codes in the dialplan with *
${HASH(SIP_CAUSE,<channel-name>)}* - This is working fine for me too.

This is the dialplan line just after the DIAL()
same => n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)}):4:3})

Ref:  http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS

Regards,
Sammy


On Tue, Feb 28, 2012 at 1:00 AM, John Millican <john at millican.us> wrote:

> Hello,
> I am using a mix of Call files and AMI telnet from a perl app to place
> calls.  I sometimes get this in the CLI:
>
>  -- Attempting call on sip/5555551234@<provider>for 1@<mycontext>:1
> (Retry 1)
> [Feb 27 13:47:07]   == Using SIP RTP CoS mark 5
> [Feb 27 13:47:07]     -- Got SIP response 503 "No Circuit Available" back
> from xxx.xxx.xxx.xxx:5060
> [Feb 27 13:47:07] > Channel SIP/<provider> was never answered.
>
> I would like to be able to capture the "Got SIP response 503 "No Circuit
> Available" back from xxx.xxx.xxx.xxx:5060"  line in a var to be used by a
> perl AGI that inserts to a mongoDB for reporting.  Is this possible?  I
> have read many articles about using hangupcause and siphangupcause but they
> do not provide the same information I believe because the call was never
> answered so hangup does not apply.
>
> TIA,
> JohnM
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/41de4e72/attachment.htm>