| Summary: | ASTERISK-13241: Incoming SIP invites don't match properly | ||
| Reporter: | Jared Smith (jsmith) | Labels: | |
| Date Opened: | 2008-12-18 09:15:44.000-0600 | Date Closed: | 2008-12-18 11:59:34.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Registration |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) logfile ( 1) sip.conf | |
| Description: | I have a Polycom phone with two separate registrations. Both "lines" on the Polycom are able to register correctly to Asterisk, and receive calls just fine. When an INVITE comes from the second line appearance on the phone into Asterisk, however, Asterisk rejects the INVITE saying that there's an authname mismatch, and gives the name of the first line appearance. This seems to me to be limited to Polycom phones at the moment, as my Linksys uses a different SIP source port for each registration. ****** STEPS TO REPRODUCE ****** 1) Use the attached sip.conf 2) Configure a Polycom phone for each of the two sip friends with "polycom" in the name. 3) Call from the second line on the Polycom into Asterisk | ||
| Comments: | By: Jared Smith (jsmith) 2008-12-18 09:16:38.000-0600 I've uploaded a logfile showing what happens, with sip debugging turned on. By: frank koster (notthematrix) 2008-12-18 10:34:48.000-0600 Same problem occoures with the siemens c6xx dect voip series. the can be called but can not make a call. the siemens can store 6 separate idetities in its memory. so that makes the bog not limited to 1 type of phone By: Jared Smith (jsmith) 2008-12-18 10:51:15.000-0600 I've tried this in Asterisk trunk, and it no longer happens there. This appears to be limited to the 1.6.1 branch at this time. By: Digium Subversion (svnbot) 2008-12-18 11:59:33.000-0600 Repository: asterisk Revision: 165606 _U branches/1.6.1/ U branches/1.6.1/channels/chan_sip.c ------------------------------------------------------------------------ r165606 | file | 2008-12-18 11:59:33 -0600 (Thu, 18 Dec 2008) | 4 lines Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers. (closes issue ASTERISK-13241) Reported by: jsmith ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=165606 | ||