| Summary: | ASTERISK-10099: hint is hanging when remote party ends call on hold (re: 0010399 | ||
| Reporter: | acennami (acennami) | Labels: | |
| Date Opened: | 2007-08-16 16:19:15 | Date Closed: | 2007-11-05 12:50:23.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Addons/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I believe this issue was supposed to be fixed, however I am running the latest release (1.4.10.1) and the issue persists. No active calls on system, however hint for particular extension still reports "Hold" and blinks BLF. Phone in question is SNOM 320 6.2.2 | ||
| Comments: | By: acennami (acennami) 2007-08-16 16:20:03 cl203-154*CLI> show channels Channel Location State Application(Data) 0 active channels 0 active calls cl203-154*CLI> show hints cl203-154*CLI> -= Registered Asterisk Dial Plan Hints =- 3099@intelecom-extensions: SIP/3099 State:Unavailable Watchers 0 3054@intelecom-extensions: SIP/3054 State:Unavailable Watchers 3 3053@intelecom-extensions: SIP/3053 State:Idle Watchers 2 3030@intelecom-extensions: SIP/3030 State:Hold Watchers 3 3028@intelecom-extensions: SIP/3028 State:Unavailable Watchers 2 3026@intelecom-extensions: SIP/3026 State:Idle Watchers 3 3024@intelecom-extensions: SIP/3024 State:Unavailable Watchers 0 3023@intelecom-extensions: SIP/3007 State:Idle Watchers 0 3022@intelecom-extensions: SIP/3022 State:Idle Watchers 2 3020@intelecom-extensions: SIP/3020 State:Unavailable Watchers 0 3017@intelecom-extensions: SIP/3017 State:Unavailable Watchers 0 3016@intelecom-extensions: SIP/3016 State:Unavailable Watchers 0 3013@intelecom-extensions: SIP/3013 State:Unavailable Watchers 1 3011@intelecom-extensions: SIP/3011 State:Idle Watchers 1 3010@intelecom-extensions: SIP/3010 State:Idle Watchers 1 3009@intelecom-extensions: SIP/3004 State:Idle Watchers 0 3008@intelecom-extensions: SIP/3007 State:Idle Watchers 1 3007@intelecom-extensions: SIP/3007 State:Idle Watchers 1 3005@intelecom-extensions: SIP/3005 State:Idle Watchers 2 3004@intelecom-extensions: SIP/3004 State:Idle Watchers 0 3066@intelecom-extensions: SIP/3066 State:Idle Watchers 1 ---------------- - 21 hints registered By: Joshua C. Colp (jcolp) 2007-08-19 20:26:21 Please provide sip.conf minus passwords, sip show subscriptions, and console output with debug of a call that causes this. By: Francesco Romano (francesco_r) 2007-09-11 02:01:26 I had yesterday the same problem on one of my servers with release 1.4SVN81952 and Grandstream phones. I had to stop and restart asterisk to resume. I'll try to reproduce in a test environment and post the necessary logs. By: Adam Long (worldlink) 2007-09-11 12:26:08 I am also having this problem but with 1.4.9 with SNOM 300 and SNOM 320 firmware 6.5.8. All Polycom extensions are working just fine or seem to be. I'm upgrading to 1.4.11 will post back any changes in behavior. By: Adam Long (worldlink) 2007-09-11 14:26:32 1.4.11 seemed to fix this for me. But now i'm getting 1 way audio issues anytime anyone either Parks a call and another station picks up or when someone does a transfer. Very odd... Here is the sip.conf for that user. [someuser1] type=friend username=someuser1 secret=xxxxxxxx callerid=Charlie <1001> mailbox=1001@aries,0@aries subscribecontext=arieshints context=aries host=dynamic disallow=all allow=ulaw nat=yes canreinvite=no dtmfmode=info qualify=10000 promiscredir=no notifyringing=yes The originating and terminating sip trunking peer also has canreinvite set to no. By: Jason Parker (jparker) 2007-09-13 11:03:50 Worldlink, please open a new report for that issue, as it is unrelated to this one. By: pkempgen (pkempgen) 2007-09-25 07:29:52 Same problem for me with Asterisk 1.4.11 $ asterisk -rx 'core show channels' | grep '100' $ $ asterisk -rx 'core show hints' | grep 100 100@to-internal-users-se: SIP/100 State:Hold Watchers 0 $ $ asterisk -rx 'dialplan show' | grep 100 '100' => hint: SIP/100 [pbx_ael] $ By: pkempgen (pkempgen) 2007-09-25 08:39:13 This may or may not be related to http://bugs.digium.com/view.php?id=10323 http://bugs.digium.com/view.php?id=10319 http://bugs.digium.com/view.php?id=10165 By: Digium Subversion (svnbot) 2007-11-05 12:45:57.000-0600 Repository: asterisk Revision: 88671 U branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r88671 | file | 2007-11-05 12:45:55 -0600 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue ASTERISK-10617) Reported by: francesco_r Tested by: blitzrage (closes issue ASTERISK-10099) Reported by: acennami ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-05 12:50:23.000-0600 Repository: asterisk Revision: 88673 _U trunk/ U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r88673 | file | 2007-11-05 12:50:22 -0600 (Mon, 05 Nov 2007) | 15 lines Merged revisions 88671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue ASTERISK-10617) Reported by: francesco_r Tested by: blitzrage (closes issue ASTERISK-10099) Reported by: acennami ........ ------------------------------------------------------------------------ | ||