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
>
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