[asterisk-users] Zombie SIP channels
on sip show channels I do get a lot of entrys like 192.168.1.47 11 07ba5a490b3 00102/0 unkn No Init: INVITE 192.168.1.47 11 19090f115b8 00102/0 unkn No Init: INVITE 192.168.1.47 11 7d8b8fde46f 00102/0 unkn No Init: INVITE How do they appear? How can they be removed? core show channels does not list them. Elmar ___ --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] zombie SIP channels after CURL cnam lookup
Can anyone suggest a reason why these channels might end up zombies?
The process is;
Call comes in via SIP into a context that appends the caller ID name as
follows;
[cnam-lookup]
exten =
_[2-9]X,1,set(CALLERID(name)=${CURL(http://cnam.provider.com/?co
mpanyId=555password=passwordnumber=${CALLERID(num)})})
exten = _[2-9]X,2,goto(subscriber-numbers|${EXTEN}|1)
the call is then sent to the context where the extension is defined.
This works well with high volume, but there are occasionally zombie
channels as a result, can not track down the cause;
Channel Location State Application(Data)
SIP/1.1.251.9-b6700 [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b6a0d [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b6ad0 [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b7dcf [EMAIL PROTECTED] Ring(None)
SIP/1.1.251.9-b675f [EMAIL PROTECTED] Ring(None)
The channels listed above have appeared in show channels for 2 days now.
I assume it was either because the CURL response was not returned, since
we are still in he context cnam-lookup and the next step is a goto.
Is there a way to set an absolute timeout for the set command and
continue in the dialplan if that timeout is exceeded, without impacting
timeouts further down the line?
The cnam response should come within 200ms.
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Re: [Asterisk-Users] Zombie SIP channels
Ok - I finally found out what was causing the ZOMBIE channels. Now follow me on this one :) It appears that if you are using a Cisco 7960 and are on a call and want to transfer the call to another extension - if you press more and Trnsfer and dial the extension and you hit the Trnsfer button again before the extension answers, a ZOMBIE channel is created. If you use BlindXfer, it does not create the ZOMBIE channel. I have now informed my client that if they want to do a Blind Transfer, to use the BlindXfer softkey instead of the Trnsfer softkey or just use the # key to do a blind transfer. Now, I am running Asterisk CVS-v1-0-11/12/04-15:32:45. I would be interested in knowing if later versions of asterisk exhibited this same behavior. Any feedback would be appreciated. Thanks, Pedro On Fri, 11 Feb 2005 08:32:43 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? http://bugs.digium.com/bug_view_page.php?bug_id=0002938 I don't think so, but who knows. The patch resolves a locking issue that may or may not be timing-source dependant. I've seen the issue occur after call transfers in scenario's where I used a few chan_local's. Do yourself a favour: - If you can, unload the ztdummy and test for a while. However, this may put the issue to sleep - but it won't solve it! - After that, load ztdummy again and apply the two lines in channel.c. Test again. Good chance the issue will be gone. Report results here :) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zombie SIP channels
Hi, -Original Message- Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? This was an issue in older versions of asterisk. It would help if you could tell us what setup you are running. If this is infact your problem too, a simple update of your asterisk to 1.0.3 or later will help. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Thanks for the feedback! Running CVS-v1-0-11/12/04 (stable) on Fedora Core 1 with Cisco 7960G's. Asterisk server is on public IP and Cisco 7960G is at client location NAT-ed behind a Cisco soho91-k9 with nine other Cisco 7960G's (each phone has registration expiring every 120 seconds). Here is excerpt from sip.conf [general] disallow=all allow=ulaw port=5060 context=incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no tos=reliability srvlookup=yes videosupport=no dtmfmode=inband nat=yes insecure=very [frontdesk] context=customer type=friend username=frontdesk secret=password host=dynamic canreinvite=no [EMAIL PROTECTED] nat=yes qualify=yes callerid=Front Desk 100 accountcode=customer amaflags=billing This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. Thanks! Pedro On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? This was an issue in older versions of asterisk. It would help if you could tell us what setup you are running. If this is infact your problem too, a simple update of your asterisk to 1.0.3 or later will help. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zombie SIP channels
Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. If my memory serves me correctly, this is the issue: http://bugs.digium.com/bug_view_page.php?bug_id=0002938 It's a two line fix, so if you want you can easily verify and apply manually so you don't have to introduce any other new code. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
What is odd is no meetme is being used. But may be related - thanks! Pedro On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. If my memory serves me correctly, this is the issue: http://bugs.digium.com/bug_view_page.php?bug_id=0002938 It's a two line fix, so if you want you can easily verify and apply manually so you don't have to introduce any other new code. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? - Pedro On Thu, 10 Feb 2005 08:50:35 -0500, Pedro [EMAIL PROTECTED] wrote: What is odd is no meetme is being used. But may be related - thanks! Pedro On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would rather not upgrade if possible has everything seems running fine. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. If my memory serves me correctly, this is the issue: http://bugs.digium.com/bug_view_page.php?bug_id=0002938 It's a two line fix, so if you want you can easily verify and apply manually so you don't have to introduce any other new code. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zombie SIP channels
Hi, -Original Message- Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? http://bugs.digium.com/bug_view_page.php?bug_id=0002938 I don't think so, but who knows. The patch resolves a locking issue that may or may not be timing-source dependant. I've seen the issue occur after call transfers in scenario's where I used a few chan_local's. Do yourself a favour: - If you can, unload the ztdummy and test for a while. However, this may put the issue to sleep - but it won't solve it! - After that, load ztdummy again and apply the two lines in channel.c. Test again. Good chance the issue will be gone. Report results here :) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zombie SIP channels
Does anyone know how to kill a zombie channel? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zombie SIP channels
I tried to send this earlier but does not look like it went through for some reason. If you get this twice - my appologies. Does anyone know how to kill a zombie channel (and why do they pop up)? Here is what I see on a show channels: -- show channels Channel (ContextExtensionPri ) State Appl. Data SIP/frontdesk-72c7 (customercontext 1 ) Up Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Pedro wrote: No one is on a call - how can I get rid of this without restarting asterisk? soft hangup TAB in Asterisk console. It'd pay to try and find out why you're getting them though. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
Thanks for the tip. They both seemed to go away on their own after a while with no action on my part. I am not sure what caused it (there is nothing in the log file). This is the first time I have seen it on any of my asterisk machines (and I have been working with asterisk for a year now). Any ideas on why a zombie sip channel would occur? Thanks in advance for any insight on this. - Pedro On Thu, 10 Feb 2005 14:57:17 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Pedro wrote: No one is on a call - how can I get rid of this without restarting asterisk? soft hangup TAB in Asterisk console. It'd pay to try and find out why you're getting them though. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zombie SIP channels
- Original Message - From: Pedro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 10, 2005 4:08 AM Subject: Re: [Asterisk-Users] Zombie SIP channels Thanks for the tip. They both seemed to go away on their own after a while with no action on my part. I am not sure what caused it (there is nothing in the log file). This is the first time I have seen it on any of my asterisk machines (and I have been working with asterisk for a year now). Any ideas on why a zombie sip channel would occur? Hi, I've spotted similar behaviour. I think that some registration process, or notifications (particularly if you put some settings for MWI on Gradnstream) causes those channels to be active for some short time. This collides with dialparties.agi (from AMP) detection of active calls to SIP clients and sometimes it sings them as being busy ... Regards, Rob. Thanks in advance for any insight on this. - Pedro On Thu, 10 Feb 2005 14:57:17 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Pedro wrote: No one is on a call - how can I get rid of this without restarting asterisk? soft hangup TAB in Asterisk console. It'd pay to try and find out why you're getting them though. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
