Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/31-1 (default 9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network 968290897 2 ) Ring Dial Zap/g2/68290897 Zap/30-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network 993841544 2 ) Ring Dial Zap/g2/93841544 Zap/29-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network 996644687 2 ) Ring Dial Zap/g2/96644687 Zap/28-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network 993871648 2 ) Ring Dial Zap/g2/93871648 Zap/27-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network 968627224 2 ) Ring Dial Zap/g2/68627224 Zap/26-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network 965627780 2 ) Ring Dial Zap/g2/65627780 Zap/25-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/24-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/23-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network 965699062 2 ) Ring Dial Zap/g2/65699062 Zap/22-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network 965676388 2 ) Ring Dial Zap/g2/65676388 Zap/21-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network 962662272 2 ) Ring Dial Zap/g2/62662272 Zap/20-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 964290118 2 ) Ring Dial Zap/g2/64290118 Zap/19-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network 965627640 2 ) Ring Dial Zap/g2/65627640 Zap/18-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network 964255575 2 ) Ring Dial Zap/g2/64255575 Zap/17-1 (default s 1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 965699062 2 ) Up Dial Zap/g2/65699062 Regards, Antonio Rabena -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040401/56e02b5b/attachment.htm
What version of Asterisk are you using.. I updated to the latest CVS yesterday and have started having the same problem.. I am busy building a new box to use from my Asterisk so will see if it is still a problem and a fresh install.. later.. Antonio Rabena wrote:> Hi, i have an asterisk box running with E100P (E1) line as PSTN > gw. Sometimes zap channels hang and i couldn't make any PSTN calls > but SIP calls are still fine. When this happens I also couldn't > restart/reload asterisk from the CLI. I have to kill the asterisk > process and run safe_asterisk again. any ideas? > > > asterisk*CLI> show channels > Channel (Context Extension Pri ) State Appl. Data > Zap/31-1 (default 9388 1 ) Dialing AppDial > (Outgoing Line) > SIP/1024-1330 (network 968290897 2 ) Ring Dial > Zap/g2/68290897 > Zap/30-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1004-bca1 (network 993841544 2 ) Ring Dial > Zap/g2/93841544 > Zap/29-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-1fa1 (network 996644687 2 ) Ring Dial > Zap/g2/96644687 > Zap/28-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-f3c0 (network 993871648 2 ) Ring Dial > Zap/g2/93871648 > Zap/27-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-aa22 (network 968627224 2 ) Ring Dial > Zap/g2/68627224 > Zap/26-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-e6e3 (network 965627780 2 ) Ring Dial > Zap/g2/65627780 > Zap/25-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-70b1 (network 963167838 2 ) Ring Dial > Zap/g2/63167838 > Zap/24-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-6e19 (network 963167838 2 ) Ring Dial > Zap/g2/63167838 > Zap/23-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-76ce (network 965699062 2 ) Ring Dial > Zap/g2/65699062 > Zap/22-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-12dd (network 965676388 2 ) Ring Dial > Zap/g2/65676388 > Zap/21-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-527d (network 962662272 2 ) Ring Dial > Zap/g2/62662272 > Zap/20-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/811586002-037a (default 964290118 2 ) Ring Dial > Zap/g2/64290118 > Zap/19-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-dc3c (network 965627640 2 ) Ring Dial > Zap/g2/65627640 > Zap/18-1 (default s 1 ) Dialing AppDial > (Outgoing Line) > SIP/1007-49ad (network 964255575 2 ) Ring Dial > Zap/g2/64255575 > Zap/17-1 (default s 1 ) Up Bridged Call > SIP/1007-de63 > SIP/1007-de63 (network 965699062 2 ) Up Dial > Zap/g2/65699062 > > > > Regards, > > > Antonio Rabena >
Im using the stable versoin 0.7.2. At 04:23 PM 4/1/2004, you wrote:>What version of Asterisk are you using.. I updated to the latest CVS >yesterday and have started having the same problem.. > >I am busy building a new box to use from my Asterisk so will see if it is >still a problem and a fresh install.. > >later..Regards, Antonio Rabena -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040401/fe178754/attachment.htm
I am having the some problem here, I had to put a asterisk restart in cron every night. I am running an E100P also, my * ver is Asterisk CVS-02/25/04-20:35:20 Thanks! Luciano -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Zap Channels Hang Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/31-1 (default 9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network 968290897 2 ) Ring Dial Zap/g2/68290897 Zap/30-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network 993841544 2 ) Ring Dial Zap/g2/93841544 Zap/29-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network 996644687 2 ) Ring Dial Zap/g2/96644687 Zap/28-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network 993871648 2 ) Ring Dial Zap/g2/93871648 Zap/27-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network 968627224 2 ) Ring Dial Zap/g2/68627224 Zap/26-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network 965627780 2 ) Ring Dial Zap/g2/65627780 Zap/25-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/24-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/23-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network 965699062 2 ) Ring Dial Zap/g2/65699062 Zap/22-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network 965676388 2 ) Ring Dial Zap/g2/65676388 Zap/21-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network 962662272 2 ) Ring Dial Zap/g2/62662272 Zap/20-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 964290118 2 ) Ring Dial Zap/g2/64290118 Zap/19-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network 965627640 2 ) Ring Dial Zap/g2/65627640 Zap/18-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network 964255575 2 ) Ring Dial Zap/g2/64255575 Zap/17-1 (default s 1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 965699062 2 ) Up Dial Zap/g2/65699062 Regards, Antonio Rabena __________ NOD32 1.700 (20040331) Information __________ This message was checked by NOD32 Antivirus System. http://www.nod32.com
This could possibly be related to Bug# 0001320 where Zap channels get stuck in a Rsrvd State. I inadvertently put the bug in Zaptel since I had upgraded to Zaptel 0.9.0 the same time I upgraded to asterisk v1-0_stable. When I rolled back to asterisk 0.7.1 with -DOLD_DSP_ROUTINES the problem went away. I'm going to try v1-0_stable with -DOLD_DSP_ROUTINES this weekend to see if the problem goes away. One bad side affect to 0.7.1 is occasional terrible echo on Zap channels. This behavior was not present in v1-0_stable. My $0.02 -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Thursday, April 01, 2004 8:38 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Zap Channels Hang Luciano, I was having the same thing happen after updating to that code...but since mine is in production I had to quickly go back to the code from two weeks ago. I know it's not a "solution"...but if you really need it back up now you might want to do that. Mark -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Zap Channels Hang I am having the some problem here, I had to put a asterisk restart in cron every night. I am running an E100P also, my * ver is Asterisk CVS-02/25/04-20:35:20 Thanks! Luciano -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Zap Channels Hang Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/31-1 (default 9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network 968290897 2 ) Ring Dial Zap/g2/68290897 Zap/30-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network 993841544 2 ) Ring Dial Zap/g2/93841544 Zap/29-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network 996644687 2 ) Ring Dial Zap/g2/96644687 Zap/28-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network 993871648 2 ) Ring Dial Zap/g2/93871648 Zap/27-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network 968627224 2 ) Ring Dial Zap/g2/68627224 Zap/26-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network 965627780 2 ) Ring Dial Zap/g2/65627780 Zap/25-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/24-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/23-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network 965699062 2 ) Ring Dial Zap/g2/65699062 Zap/22-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network 965676388 2 ) Ring Dial Zap/g2/65676388 Zap/21-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network 962662272 2 ) Ring Dial Zap/g2/62662272 Zap/20-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 964290118 2 ) Ring Dial Zap/g2/64290118 Zap/19-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network 965627640 2 ) Ring Dial Zap/g2/65627640 Zap/18-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network 964255575 2 ) Ring Dial Zap/g2/64255575 Zap/17-1 (default s 1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 965699062 2 ) Up Dial Zap/g2/65699062 Regards, Antonio Rabena __________ NOD32 1.700 (20040331) Information __________ This message was checked by NOD32 Antivirus System. http://www.nod32.com _______________________________________________ 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
our cvs is 02/25/04 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Juan J. Sierralta P. Sent: Thursday, April 01, 2004 11:56 AM To: Asterisk Users Subject: Re: [Asterisk-Users] Zap Channels Hang On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote:> Hi, > > I have same problem with zap channels. I have E100P installed on myasterisk> box and I worked with CVS-02/22/04-16:30:20 and everything worked well(with> Zap channels). I update asterisk to new cvs 2 days ago and incoming zap > calls starts hanging. > I have mgcp extensions defined in my extensions.conf and I see that if > voicemail is enabled for extension and there are two concurent call (from > Zap) to this extension, second call to voicemail are hanging in asterisk > after user from Zap side hangs up. If there are no voicemail for extension > the call are not hanging at all. May be these information will be helpfull > to fix this bug.I noted the same problems with CVS from 03/30/2004 when incoming calls were sent to voicemail. Anyway I had to roll back to 03/05 since last Zaptel was giving me yellow alarms con my TE410P on a E1 PRI. -- Juanjo sin .sig _______________________________________________ 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