Relaying SIP Reason, cause and text between calls

Asterisk Asterisk SIP
1

We all know how cute is asterisk when you dial two extensions like Dial(SIP/100&SIP/200) and one of the extensions answers the call: the other extension gets a SIP CANCEL with Reason: SIP;cause=200;text=“Call completed elsewhere” and the “missed call” message doesn’t appear on the other phone.

Unfortunately, the SIP cause is not being relayed when there are multiple calls in a chain. Let’s make an example. If you have three asterisk servers, name A, B and C. A is processing the call, while B and C have one extension connected. A is calling B and C using a command like Dial(SIP/A&SIP/B) and B calls extension 100 with Dial(SIP/100) and C calls extension 200 with Dial(SIP/200).

      ----- B ---- 100
    /
A   
    \
      ----- C ---- 200

If extension 200 answers the call, A sends a CANCEL to B with the reason “Call completed elsewhere”, but B will not send the same reason to Extension 100, so extension 100 will see a missed call.

Is there any way to have Extension 100 to receive the “Call completed elsewhere” when applicable?

2

Hello,
I’ve got exactly the same problem: I need a SIP CANCEL with Reason to pass through Asterisk “C” (as described by Idardini). Unfortunately, CANCEL is relayed but reason is lost.

Is there any way to acheive this?

Thanks,
Petr

3

Hi,
Did you manage to resolve your issue? I have exactly the same problem

4

You shouldn’t have the same problem as you should be using chan_pjsip, but the OP was using chan_sip.

However the earlier posting is wrong in suggesting the CANCEL is relayed. Asterisk is a back to back user agent, not a SIP proxy. What actually happens is that the cancel results in an AST_CONTROL_HANGUP message, internally, with the nearest ISDN cause code to the original SIP one, and that causes app_dial, to issue ast_hangup() to the outgoing channel, with a copy of the ISDN type hangup cause.

5

Right I’m using chan_pjsip but after receiving CANCEL with header
SIP ;cause=200 ;text=“Call completed elsewhere”

there are logs:
chan_pjsip.c:2523 hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP)
chan_pjsip.c:2621 chan_pjsip_hangup: Cause: 0
app_dial.c:3316 dial_exec_full: Exiting with DIALSTATUS=CANCEL.
chan_pjsip.c:2621 chan_pjsip_hangup: Cause: 500

in my dialplan (hangup_handler_push) I’m printing ${HANGUPCAUSE} and it’s 127. Is there any way to get info it was CANCEL with “Call completed elsewhere” ?

6

The text portion is not inspected by either chan_sip or chan_pjsip, so that’s not supported.