Summary: | ASTERISK-01910: Make RFC3581 support optional | ||
Reporter: | Ryan Courtnage (rcourtna) | Labels: | |
Date Opened: | 2004-06-28 12:00:00 | Date Closed: | 2008-01-15 15:01:04.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | ASTERISK-1837 introduced support for RFC3581 (adds rport parameter to Via field in SIP header). Unfortunately, this breaks some clients that do not support this (ie: Uniden's UIP200). I'd like to request that RFC3581 be made an option, either globally, or on a 'per client' basis in sip.conf. | ||
Comments: | By: Mark Spencer (markster) 2004-06-28 14:53:28 How does it "break" a client? By: Ryan Courtnage (rcourtna) 2004-06-28 15:02:45 sorry, should have been more descriptive. The UIP200 sip phone will not respond to INVITE requests that contain ';rport' in the Via field. ie: If you try to call a Uniden phone, the phone will not ring. If I remove ';rport' from the via in chan_sip.c, then all will work again. By: Mark Spencer (markster) 2004-06-28 19:38:52 You should now be able to use "nat=never" to to specifically force the rport to be omitted. By: Mark Spencer (markster) 2004-06-28 19:39:15 Please confirm it fixes it for you as soon as humanly possible as I am leaving for Europe very soon. By: Ryan Courtnage (rcourtna) 2004-06-28 20:12:49 Markster, Yes, this fixes the problem. Thanks! By: Digium Subversion (svnbot) 2008-01-15 15:01:04.000-0600 Repository: asterisk Revision: 3335 U trunk/channels/chan_sip.c U trunk/configs/sip.conf.sample ------------------------------------------------------------------------ r3335 | markster | 2008-01-15 15:01:03 -0600 (Tue, 15 Jan 2008) | 2 lines Allow nat=never mode to work around buggy UNIDEN UIP200 firmware (bug ASTERISK-1910) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=3335 |