Re: [asterisk-users] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? -Stephen- ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Wed, 2007-05-30 at 08:56 -0600, Stephen Bosch wrote: Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Wed, 2007-05-30 at 08:56 -0600, Stephen Bosch wrote: Carlos Chavez wrote: On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. I'm suspicious of the Zap channel in the off-hook state. It should on-hook when on-hook and off-hook when in use. Is that channel still in off-hook? You say you made no changes, it just stopped working. Did anything *else* change around the time this problem appeared? Did someone move a device, or did you update a driver? No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. Sounds like you need to stop obsessing over the Zap ports. Your problem is with the SIP phones. Can two SIP phones on that system call each other? ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Wed, 2007-05-30 at 14:49 -0500, Eric ManxPower Wieling wrote: No changes have been made in a while. The customer is very particular and very difficult to deal with so we NEVER make any changes unless it is absolutely necessary. Today I made several more tests and things are getting weirder. First I tried to connect the Vonage ATA to another port on the card just to confirm that it was not the port who was the problem. I has the exact same problem. I cannot call any SIP phone, they all give me Auto-Congestion. If I change the dialplan to dial an analog phone it goes through without a hitch, the problem only presents itself with the SIP phones. Stranger still is that I can use a SIP phone to dial to the Vonage line, only incoming calls have a problem. Sounds like you need to stop obsessing over the Zap ports. Your problem is with the SIP phones. Can two SIP phones on that system call each other? Everything else in the system works, all sip phones can call each other and the PSTN. They have a GSM adapter on the same card and they can place and receive calls. Only calls coming from the Vonage ATA have this problem. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Wed, 2007-05-30 at 14:49 -0500, Eric ManxPower Wieling wrote: Can two SIP phones on that system call each other? Everything else in the system works, all sip phones can call each other and the PSTN. They have a GSM adapter on the same card and they can place and receive calls. Only calls coming from the Vonage ATA have this problem. I missed the first part of the thread. Can you paste the CLI output of a successful call (SIP phone to SIP phone) and an unsuccessful call (Zap to SIP)? ___ --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] Problem on incoming call from Zap channel to SIP phones...
I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP phones. The port is configured to answer, dial SIP/603 and if that line is busy it will dial to SIP/604. Here is the output from the CLI: -- Starting simple switch on 'Zap/40-1' -- Executing Answer(Zap/40-1, ) in new stack -- Executing Set(Zap/40-1, TIMEOUT(response)=5) in new stack -- Response timeout set to 5 -- Executing Dial(Zap/40-1, SIP/603|20) in new stack -- Called 603 May 29 17:15:35 NOTICE[28703]: chan_sip.c:2012 auto_congest: Auto-congesting SIP/603-0080d790 -- SIP/603-0080d790 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) As you can see I get the message that says that the phone is busy even when it is not. This happens no matter which SIP phone I try to dial on the server. I can dial from any other phone to the same extension without any problems. Only calls coming from the Vonage line have this problem. I can make outgoing calls using that same line. This was working until mid day today and there were no changes made to the configuration until after the problem was detected. The only thing I notice is that if I do a zap show channel 40 it tells me Hookstate (FXS only): Offhook even when the line is not in use. The other port connected to the GSM adapter says Onhook. Also, when Asterisk answers the call from Vonage I hear a loud click on the phone and after it has tried both extensions I can hear the Voicemail message play. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Tue, May 29, 2007 at 05:22:45PM -0500, Carlos Chavez wrote: I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP phones. The port is configured to answer, dial SIP/603 and if that line is busy it will dial to SIP/604. Here is the output from the CLI: -- Starting simple switch on 'Zap/40-1' -- Executing Answer(Zap/40-1, ) in new stack -- Executing Set(Zap/40-1, TIMEOUT(response)=5) in new stack -- Response timeout set to 5 -- Executing Dial(Zap/40-1, SIP/603|20) in new stack -- Called 603 May 29 17:15:35 NOTICE[28703]: chan_sip.c:2012 auto_congest: Auto-congesting SIP/603-0080d790 -- SIP/603-0080d790 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) As you can see I get the message that says that the phone is busy even when it is not. This happens no matter which SIP phone I try to dial on the server. I can dial from any other phone to the same extension without any problems. Only calls coming from the Vonage line have this problem. I can make outgoing calls using that same line. This was working until mid day today and there were no changes made to the configuration until after the problem was detected. The only thing I notice is that if I do a zap show channel 40 it tells me Hookstate (FXS only): Offhook even when the line is not in use. The other port connected to the GSM adapter says Onhook. Also, when Asterisk answers the call from Vonage I hear a loud click on the phone and after it has tried both extensions I can hear the Voicemail message play. What is the output of: show channels Any chance that there is actually another call that keeps that channel busy? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote: What is the output of: show channels Any chance that there is actually another call that keeps that channel busy? No, the line is not busy with another call. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote: On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote: What is the output of: show channels Any chance that there is actually another call that keeps that channel busy? No, the line is not busy with another call. In fact at this moment there are 0 active calls in the system. The phones are all Aastra 9133i which can support up to 9 calls. I have restarted the service, rebooted the server and even upgraded to the latest zaptel drivers and made sure CentOS is patched and up to date. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Tue, 2007-05-29 at 19:14 -0500, Carlos Chavez wrote: On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote: On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote: What is the output of: show channels Any chance that there is actually another call that keeps that channel busy? No, the line is not busy with another call. In fact at this moment there are 0 active calls in the system. The phones are all Aastra 9133i which can support up to 9 calls. I have restarted the service, rebooted the server and even upgraded to the latest zaptel drivers and made sure CentOS is patched and up to date. I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Problem on incoming call from Zap channel to SIP phones...
Carlos Chavez wrote: On Tue, 2007-05-29 at 19:14 -0500, Carlos Chavez wrote: On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote: On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote: What is the output of: show channels Any chance that there is actually another call that keeps that channel busy? No, the line is not busy with another call. In fact at this moment there are 0 active calls in the system. The phones are all Aastra 9133i which can support up to 9 calls. I have restarted the service, rebooted the server and even upgraded to the latest zaptel drivers and made sure CentOS is patched and up to date. I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. sip show peers will list the IP address of the phones. If it is not listed then you have a problem with your phones not registering to Asterisk. ___ --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] Problem on incoming call from Zap channel to SIP phones...
On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote I just made another test by dialing to a Zap channel instead of a SIP phone and the call goes through without any problem. It is just when you try to dial to a SIP phone that you get the auto-congestion message. All other phones in the system are working properly, they are all registered and you can send and receive calls from anywhere except that zap channel. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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
