Summary: | ASTERISK-08469: Replace not working properly? | ||
Reporter: | deti (deti) | Labels: | |
Date Opened: | 2007-01-03 09:41:58.000-0600 | Date Closed: | 2007-10-30 10:30:36 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) replaces.log | |
Description: | Tried to replace a call and got: 'Supervised transfer attempted to replace non-ringing or active call id(...)' There is a check in chan_sip.c near line 13342: if (!error && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) I guess this should be something like if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_UP ) But I still run into trouble with this change: The call is not being cancelled on the RINGING device and therefor device state is not being updated properly. Am I doing something wrong? Any clues? | ||
Comments: | By: Serge Vecher (serge-v) 2007-01-03 09:45:19.000-0600 Deti: as per bug guidelines, can you please attach a sip debug with and without your change? 1) Prepare test environment (reduce the amount of unrelated traffic on the server); 2) Make sure your logger.conf has the following line: console => notice,warning,error,debug 3) restart Asterisk with the following command: 'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt' 4) Enable SIP transaction logging with the following CLI commands: set debug 4 set verbose 4 sip debug 5) Trim startup information and attach verbosedebug.txt to the issue. By: deti (deti) 2007-01-03 10:37:58.000-0600 This is how it looks before: *CLI> sip show subscriptions Peer User Call ID Extension Last state Type Mailbox xxx 71034554-B 3c2674624e2 71034577@db-ext Idle dialog-info+xml <none> 1 active SIP subscription *CLI> core show hints -= Registered Asterisk Dial Plan Hints =- 71034577@db-extensions : SIP/71034567-U State:Idle Watchers 1 Now I call extension 71034577 Peer User Call ID Extension Last state Type Mailbox xxx 71034554-B 3c26700b41e 71034577@db-ext Ringing dialog-info+xml <none> *CLI> core show hints -= Registered Asterisk Dial Plan Hints =- 71034577@db-extensions : SIP/71034567-U State:Ringing Watchers 1 See log what happens when I do a replace on a call to extension 71034577. Then after hanging up the call you still can see: core show hints -= Registered Asterisk Dial Plan Hints =- 71034577@db-extensions : SIP/71034567-U State:Ringing Watchers 0 By: Serge Vecher (serge-v) 2007-03-13 10:05:44 deti: is this still an issue with 1.4.1? If so, can we please have a new verbosedebug.txt? By: deti (deti) 2007-03-13 10:14:48 Sorry, can't test right now. Let's close the bug for now if I'm the only one who was affected by that. By: Serge Vecher (serge-v) 2007-03-13 11:32:07 timrobbins: are you able to test 1.4.1? By: Olle Johansson (oej) 2007-03-14 07:54:44 We don't want to be in RING mode, we want the call to be in RINGING mode, so your change is wrong. I'll take a log at your debug files later (occupied at a conference). By: Olle Johansson (oej) 2007-05-15 15:33:03 well, I'm sorry. The conference wasn't a one-month conference. But there has been a lot to do and I am way behind in the bug tracker. Will try to get this checked during the dev meeting next week. By: Gregory Hinton Nietsky (irroot) 2007-07-07 05:01:28 hi there this is a problem still see http://bugs.digium.com/view.php?id=10143 and i also have patched sip to allow both RING and RINGING. By: Gregory Hinton Nietsky (irroot) 2007-08-08 10:15:22 this can probably be closed see ASTERISK-1008143 it is related/duplicate and resolves this issue By: Joshua C. Colp (jcolp) 2007-10-30 10:30:35 Closed as it should already be fixed by change in 10143. |