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 16:02:25.000-0600 | Date Closed: | 2008-02-29 12:31:49.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/NewFeature |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
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: SIP_AUTH_NAME SIP_AUTH_SECRET SIP_AUTH_MD5SECRET | ||
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. -Maxim 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: SIP/extension[:password:[md5secret[:authname]]]@peer We've left the second form unchanged: SIP/peer[/extension] Hopefully it's sifficient. -Maxim 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. -Maxim 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 Patches: chan_sip.c-trunk.diff uploaded by sobomax (license 359) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=99521 By: Maxim Sobolev (sobomax) 2008-01-23 11:59:24.000-0600 I've uploaded the sip.conf.sample diff. -Maxim 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 Patches: dialstring_doc.diff uploaded by sobomax (license 359) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=105378 |