Re: [asterisk-users] Force SIP hang up.
soft hangup channelname -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Casey Sent: Wednesday, July 18, 2007 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Force SIP hang up. Is there a way to hang up on a sip channel. One of my phones is saying it's busy while it's not (even after rebooting it). I logged into asterisk, and did a sip show channel 232, and sure enough it thinks it's on a call. How can I force it to close? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lines Not being Hung UP Major
Can't help you with the cause but I can tell you that you can use the soft hangup command to kill those channels instead of restarting. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Thursday, July 12, 2007 3:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Lines Not being Hung UP Major Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
use the safe_asterisk script it will restart asterisk if it crashes and it enables core dumps (your core size limit is probably set to 0 when you start asterisk). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Tuesday, June 26, 2007 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [asterisk-users] kore dump I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it's running, no harm done, and if it crashes, the cron job will make sure that it's started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.5
Turn off debug From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Friday, June 22, 2007 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: [asterisk-users] 1.4.5 I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED] Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phantom Calls
Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, June 18, 2007 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phantom Calls Stephen Bosch wrote: Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP RealTime Friends
try enabling rtcachefriends -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Friday, May 04, 2007 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP RealTime Friends Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
Perhaps the context in sip.conf doesn't match the context in the dial plan. From: [EMAIL PROTECTED] on behalf of Mr. Jones Sent: Fri 8/11/2006 2:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] cmd DIAL - Who picked up the call?
DIALEDPEERNUMBER contains the exact peer spec for the peer that picked up. You can use that. From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter Sent: Wed 8/2/2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] cmd DIAL - Who picked up the call? On Wednesday, August 02, 2006 6:49 AM Eric ManxPower Wieling wrote: Zap/10-43 would indicate that this is the 43rd call (call waiting) on channel 10. Obviously this would have to be removed to do it the way you want. Obviously. :-) Or we find another solution for the problem/challange... Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] cmd DIAL - Who picked up the call?
No idea, but DIALEDPEERNAME should contain the same value as BRIDGEPEER. Try that. The only difference is that BRIDGEPEER is set slightly later (when the call is bridged). From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter Sent: Wed 8/2/2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] cmd DIAL - Who picked up the call? On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote: DIALEDPEERNUMBER contains the exact peer spec for the peer that picked up. You can use that. Consider yourself my hero of the day! That looks VERY promising. It does not show the technology so Dial(SIP/phone_200Zap/g2/13,,M(getchannel)) will return either phone_200 or g2/13 but I will find a solution for that as well I suppose!!! Any idea why BRIDGEPEER is empty here all the time? Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] more voicemail frustrations (was: realtimevoicemail)
[EMAIL PROTECTED] wrote: Vadim Berezniker wrote: That's not a solution, but just a workaround. 1.2.1 has a bug where it always uses an empty context when searching for a mailbox when using realtime config. At around line 546 of apps/app_voicemail.c there is a line that says var = ast_load_realtime(voicemail, mailbox, mailbox, context, retval-context, NULL); Change it to var = ast_load_realtime(voicemail, mailbox, mailbox, context, context, NULL); Then recompile and contexts will work. looks like they've fixed it in 1.2.2. however. i switched over to use realtime SIP. now the voicemail light doesn't work. also has anyone use the MailboxExists() function in dial plan? seems like no matter what i do. it'll just just execute the next proprity. :( Yes, they fixed it in 1.2.2. They also fixed a SQL quote bug (that affected MSSQL and maybe others but not mysql) There's still an outstanding bug on the usage of the wrong SQL data type in an ODBC call (but once again it doesn't affect mysql). I didn't have problems using the voicemail indicator with realtime. Are you getting any errors in the console? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection pooling
I wrote a connection pooling patch because asterisk is not usable with MSSQL without it. If you're using, or would like to use, MSSQL I recommend you to check it out. http://bugs.digium.com/file_download.php?file_id=8809type=bug Just so you know, this is a diff against 1.2.1 and it's been only tested with 1.2.1 (because that's what we are using in production). To enable pooling, you need to edit your res_odbc.conf file and add pooled = yes poolsize = 10 To the entries you want to be pooled This also adds an additional command odbc showpool that shows detailed information about the connection pools. We've been running with this patch on two machines and it seems to be working just fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] realtime voicemail
[EMAIL PROTECTED] wrote: On Monday 16 January 2006 09:02, [EMAIL PROTECTED] wrote: Put in voicemail.conf searchcontexts=yes and do not forget to stop and start *. Reload may not do. benchev That's not a solution, but just a workaround. 1.2.1 has a bug where it always uses an empty context when searching for a mailbox when using realtime config. At around line 546 of apps/app_voicemail.c there is a line that says var = ast_load_realtime(voicemail, mailbox, mailbox, context, retval-context, NULL); Change it to var = ast_load_realtime(voicemail, mailbox, mailbox, context, context, NULL); Then recompile and contexts will work. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
