| Summary: | ASTERISK-05779: Calls that come in over Zap channel to SIP phone cannot be "unholded" | ||
| Reporter: | nenadr (nenadr) | Labels: | |
| Date Opened: | 2005-12-05 07:28:49.000-0600 | Date Closed: | 2011-06-07 14:10:35 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_zap |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | When a call comes in over a any kind of Zap channel (ISDN/PRI X100P or TDM400) and it get answered by SIP phone, and then put to hold by a SIP phone it cannot be un-holded. Tried with Cisco 7912G SIP phone and Swisvoice IP10S SIP phone. Tried with mpg123 based music on hold and file based music on hold. In all cases I have same behaviour as I have explained. SIP phones have canreinvite=no in peer config, and Dial command has t and T flags to keep media path through the Asterisk. | ||
| Comments: | By: Russell Bryant (russell) 2005-12-05 14:53:48.000-0600 Please provide 'sip debug' output as well as other console output when you recreate this problem. By: nenadr (nenadr) 2005-12-07 13:34:04.000-0600 sorry for the delay, I was out of town for a 2 days.... I will test it with 1.2.1 tomorrow and report back sip debug and console logs. By: nenadr (nenadr) 2005-12-08 13:05:24.000-0600 Problem was caused by chan_sip part of T38 patch from issue ASTERISK-4957. Problem is that INVITE from SIP phone with in c=IN IP4 0.0.0.0 as a hold request, don't get 200 OK at all. Since I'm trying to bring T38 closer *, I'll look into this problem with a next patch for issue ASTERISK-4957. You can close this one. Sorry for inconvinience. By: Russell Bryant (russell) 2005-12-08 13:28:28.000-0600 Thank you for your help in testing patches! By: Russell Bryant (russell) 2005-12-08 13:28:45.000-0600 Thank you for your help in testing patches! | ||