Summary:ASTERISK-11071: [patch] Allow AGI script writer to set username/password for the outgoing SIP call leg
Reporter:Maxim Sobolev (sobomax)Labels:
Date Opened:2007-12-17 16:02:25.000-0600Date Closed:2008-02-29 12:31:49.000-0600
Versions:Frequency of
Environment:Attachments:( 0) chan_sip.c.diff2
( 1) chan_sip.c-trunk.diff
( 2) dialstring_doc.diff
Description:Currently there is no way to specify username/password for the outgoing call leg from the AGI script (i.e. DIAL SIP/...). The following patch adds such capability by introducing three new channel variables that the script can set before calling DIAL:

Comments:By: Olle Johansson (oej) 2007-12-18 02:14:40.000-0600

I think the proper place for this information is in the dial string someplace, not in channel variables. IAX2 already has that.

The problem is the confusion between the username part of the outbound URI and the auth username, but we already have syntax for that in the register= statement in sip.conf.

By: Olle Johansson (oej) 2007-12-18 02:15:07.000-0600

All new features need to be patches for svn trunk, not a released version.

By: Maxim Sobolev (sobomax) 2007-12-18 02:56:01.000-0600

Olle, you are right, thank you for suggestion. We will re-implement this feature as DIAL string and submit it as a follow-up.


By: Maxim Sobolev (sobomax) 2007-12-19 12:14:13.000-0600

Please find updated patch attached. It allows providing authentication in the following format:


We've left the second form unchanged:


Hopefully it's sifficient.


By: Olle Johansson (oej) 2007-12-19 12:31:58.000-0600

Well, only supporting one of three different choices is not good enough, but I'll take a look at what I can do... I messed up for you with that commit today...

By: Maxim Sobolev (sobomax) 2007-12-19 14:14:28.000-0600

Well, arguably this is a new functionality so that we can avoid ambiguity and provide only single way to do it. However, if you don't feel so feel free to extend the patch.

In any case thanks for the prompt reply.


By: Olle Johansson (oej) 2008-01-22 09:18:57.000-0600

Back again. Thanks for the reminders :-)

By: Olle Johansson (oej) 2008-01-22 09:20:37.000-0600

Missing any form of documentation. Please add to configs/sip.conf.sample

By: Digium Subversion (svnbot) 2008-01-22 09:24:47.000-0600

Repository: asterisk
Revision: 99521

U   trunk/channels/chan_sip.c

r99521 | oej | 2008-01-22 09:24:44 -0600 (Tue, 22 Jan 2008) | 9 lines

Add authentication options to the SIP dialstring.
Documentation follows separately

(issue ASTERISK-11071)
Reported by: sobomax
     chan_sip.c-trunk.diff uploaded by sobomax (license 359)



By: Maxim Sobolev (sobomax) 2008-01-23 11:59:24.000-0600

I've uploaded the sip.conf.sample diff.


By: Digium Subversion (svnbot) 2008-02-29 12:31:12.000-0600

Repository: asterisk
Revision: 105378

U   trunk/configs/sip.conf.sample

r105378 | file | 2008-02-29 12:31:05 -0600 (Fri, 29 Feb 2008) | 6 lines

Add documentation for setting username/password in SIP dial string.
(closes issue ASTERISK-11071)
Reported by: sobomax
     dialstring_doc.diff uploaded by sobomax (license 359)