[asterisk-users] SIP Diversion RDNIS - how to get reason parameter?
Hi out there
To play the correct announcement in app_voicemail I whould be able to read the
SIP Diversion Reason which ist sent by another PBX:
Invite contains:
Diversion: sip:+41315995003@157.161.10.190;reason=no-
answer;privacy=off;counter=1
Asterisk Logs:
RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1)
From what I see in the source of chan_sip the variable ${SIPDIVERSIONREASON}
should be set, but it is empty...
Also ${PRIDIVERSIONREASON} is empty...
I'm using: Asterisk 1.6.2.5-0ubuntu1.3
Any hints?
-Benoit-
--
_
-- 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
Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?
2011/5/20 Benoit Panizzon benoit.paniz...@imp.ch Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Are those PBXs directly connected to each other through a SIP trunk ? -- _ -- 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
Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?
Hi Olivier Are those PBXs directly connected to each other through a SIP trunk ? Yes, and the reason is definitely transmitted in the SIP header and also parsed by asterisk and displayed in debug output. After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is just put in a temporary variable __SIPDIVERSIONREASON but not in a variable useable in the dialplan. Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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
Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?
On 11-05-20 10:39 AM, Benoit Panizzon wrote: After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is just put in a temporary variable __SIPDIVERSIONREASON but not in a variable useable in the dialplan. You could double check by using DumpChan() to see what channel variables are available for you throughout the dialplan flow. Also check the CHANNEL() and SIP*() functions to see if there is anything there that may be of use. Leif. -- _ -- 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
