Re: [asterisk-users] Zap channels stuck...
On Fri, Aug 08, 2008 at 10:16:31AM -0500, Carlos Chavez wrote: My office Asterisk box has a TDM04B card for three land lines and a GSM gateway. I have noticed that the Zap channels get stuck a couple times a week and I have to restart Asterisk to clear them. Here is what I see in the console: Connected to Asterisk 1.4.21.2 currently running on pbxoficina (pid = 18948) Verbosity is at least 3 pbxoficina*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo Zap/1-1 macro-stdexten s 1 Up Dial SIP/2001|25|Ww5554801230 general (None) Zap/2-1 macro-stdexten s 1 Up Dial SIP/2001|25|Ww5554801230 general (None) 2 active channels 28 active calls I have upgraded Asterisk and Zaptel to the latest stable version but I still have the same problem. The other strange thing is that if I do a service asterisk stop it does kill the malfunctioning Asterisk process but I can see that it restarts immediately which is not supposed to happen. This behavior is only present when there are stuck channels. When everything is ok it will properly stop asterisk and safe_asterisk. Just the obvious question: you have tried soft hangup, right? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels stuck...
On Fri, 2008-08-08 at 23:00 +0300, Tzafrir Cohen wrote: On Fri, Aug 08, 2008 at 10:16:31AM -0500, Carlos Chavez wrote: My office Asterisk box has a TDM04B card for three land lines and a GSM gateway. I have noticed that the Zap channels get stuck a couple times a week and I have to restart Asterisk to clear them. Here is what I see in the console: Connected to Asterisk 1.4.21.2 currently running on pbxoficina (pid = 18948) Verbosity is at least 3 pbxoficina*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo Zap/1-1 macro-stdexten s 1 Up Dial SIP/2001|25|Ww5554801230 general (None) Zap/2-1 macro-stdexten s 1 Up Dial SIP/2001|25|Ww5554801230 general (None) 2 active channels 28 active calls I have upgraded Asterisk and Zaptel to the latest stable version but I still have the same problem. The other strange thing is that if I do a service asterisk stop it does kill the malfunctioning Asterisk process but I can see that it restarts immediately which is not supposed to happen. This behavior is only present when there are stuck channels. When everything is ok it will properly stop asterisk and safe_asterisk. Just the obvious question: you have tried soft hangup, right? Yes but the channels are still there after the command. -- 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels state
You can try asterisk -rx core show channels and parse to output From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo A Gonzalez Sent: Friday, June 06, 2008 12:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zap channels state Hello people! I want to know if is there a shell, php script that show me which channels on a PRI line are onhook/offhook? Thanks for any help. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- 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] Zap Channels Collide (Incoming Outgoing)
Eric Wieling wrote: People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. I agree. But I think that comment is incredibly funny. I'd like to re-write it for about 20 years ago... (and some even today) People that try to wing it and install Networks when they don't know networking just gives people a bad impression of Servers and Computers in general. People = Telco Guys Oh, yes. I saw an entire Cat 5 network on punch blocks one time! Everybody needs to learn the other side before getting involved. Bill ___ -- 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] Zap Channels Collide (Incoming Outgoing)
There are krone blocks designed for CAT5, and I've seen them in use as well. However, there's no way I'd be using them for today's networks. /Especially/ having seen one of these krone blocks used to double-punch two network ports together. Bill Andersen wrote: Oh, yes. I saw an entire Cat 5 network on punch blocks one time! Everybody needs to learn the other side before getting involved. ___ -- 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] Zap Channels Collide (Incoming Outgoing)
Al Baker wrote: I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. Yep, unbelievable. This is the reason most PBXs are ground start, is there Zap hardware that does ground start? I really never looked. At least the outbound call won't go out on a line with a incoming call if they had ground start. -Ron ___ -- 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] Zap Channels Collide (Incoming Outgoing)
RE Kushner List Account wrote: Al Baker wrote: I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. Yep, unbelievable. This is the reason most PBXs are ground start, is there Zap hardware that does ground start? I really never looked. At least the outbound call won't go out on a line with a incoming call if they had ground start. I don't think the analog cards support anything except FXOLS and FXOKS, the newer 2400 and 800 analog cards might support this. I believe it is a driver issue rather than a hardware issue. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Zap Channels Collide (Incoming Outgoing)
I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ regards, Drew Al Baker wrote: I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. C. Chad Wallace wrote: At 5:22 PM on 08 May 2008, Forrest Beck wrote: I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? This is known in the telephony world as glare, and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] Zap Channels Collide (Incoming Outgoing)
Drew Gibson wrote: I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Zap Channels Collide (Incoming Outgoing)
Eric Wieling wrote: Drew Gibson wrote: I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. but how else do they learn? -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] Zap Channels Collide (Incoming Outgoing)
Drew Gibson wrote: Eric Wieling wrote: Drew Gibson wrote: I think the scary thing is that, for most people, basic knowledge of telephony was almost impossible to come by outside the opaque and secretive world of telco. That is until Asterisk came along! Perhaps there should be a regulatory requirement to read The Future of Telephony, cover to cover, before installing any Asterisk system! :-) http://www.asteriskdocs.org/ People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. but how else do they learn? Books are one of the best resources, the Wiki is not *too* bad when it comes to general telecom stuff. You can also build prototype systems. No, Asterisk did not suddenly unleash the gates of knowledge in telecom. All that information was available before Asterisk. What was not available was info on the specific inner workings of traditional PBXs. Asterisk and Digium did reduce the hardware cost of building a PBX. Traidional telecom is actually fairly simple if you compare it with IP PSTN/IP PBXs. With an IP PBX like Asterisk you need to understand telecom, IP networking (including routing, NAT, ports), Linux, as well as Asterisk itself. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Zap Channels Collide (Incoming Outgoing)
At 5:22 PM on 08 May 2008, Forrest Beck wrote: I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? This is known in the telephony world as glare, and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #19: If you're interested in building packages from source, you should consider installing the apt-src package. ___ -- 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] Zap Channels Collide (Incoming Outgoing)
I know that everyone has gaps in their knowledge, but I am just staggered that systems are being sold/deployed with such fundamental TELCO workings not being understood. Frightening. C. Chad Wallace wrote: At 5:22 PM on 08 May 2008, Forrest Beck wrote: I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? This is known in the telephony world as glare, and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... ___ -- 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] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
Steve Totaro wrote: My question is does ANYONE do ANY testing on these releases? It would seem that this bug is so paramount to the purpose of the code that had anyone taken a MINUTE to TEST, it would have been discovered IMMEDIATELY. Not if you already had a zaptel udev rules script installed on the system that's used as the test machine. This was a regression do to recent Makefile changes. A test for this problem has now been added to our pre-release regression testing. Matthew Fredrickson sigh. Thanks, Steve Totaro On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sean Bright to Asterisk show details 4:47 PM (15 hours ago) There is a bug in 'make install' in Zaptel 1.4.10 that causes the devices to not be installed correctly. You can either install 1.4.9 or wait for 1.4.11 to be released. On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o. [EMAIL PROTECTED] wrote: Hi list! I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linux with installed digium packets 1. Asterisk 1.4.19 2. Zaptel 1.4.10 3. Libpri 1.4.3 My Digium hardware is [EMAIL PROTECTED] ~]# zaptel_hardware pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card The problem is the asterisk doesn't recognize the Zap channels at all. The error is No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) and there is the original output form Astersik console: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20) in new stack [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type registered for 'Zap' [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new stack == Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90' And everything was working quite fine when I was on asterisk 1.2.13, previously installed on this very same server, same Digium card etc. The configurations are totaly the same, also. What could be the resolution of this problem? ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- 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] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
Sean Bright to Asterisk
show details 4:47 PM (15 hours ago)
There is a bug in 'make install' in Zaptel 1.4.10 that causes the
devices to not be installed correctly. You can either install 1.4.9 or
wait for 1.4.11 to be released.
On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o.
[EMAIL PROTECTED] wrote:
Hi list!
I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21
EST 2007 i686 i686 i386 GNU/Linux
with installed digium packets
1. Asterisk 1.4.19
2. Zaptel 1.4.10
3. Libpri 1.4.3
My Digium hardware is
[EMAIL PROTECTED] ~]# zaptel_hardware
pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I
...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card
The problem is the asterisk doesn't recognize the Zap channels at all. The
error is No channel type registered for 'Zap'
and Unable to create channel of type 'Zap' (cause 66 - Channel not
implemented) and there is the original output form Astersik console:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20)
in new
stack
[Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type
registered for 'Zap'
[Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'Zap' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new
stack
== Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90'
And everything was working quite fine when I was on asterisk 1.2.13,
previously installed on this very same server, same Digium card etc.
The configurations are totaly the same, also.
What could be the resolution of this problem?
Here are my configs
[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
fxsks=1
fxsks=2
fxols=3
fxols=4
[EMAIL PROTECTED] ~]# cat /etc/asterisk/zapata.conf
[channels]
context=incoming
callerid=yes
hidecallerid=no
imidiate=no
context=incoming
signalling=fxs_ks
echocancel=yes
group=1
channel = 1
channel = 2
context=local
signalling=fxo_ks
echocancel=yes
group=2
channel = 3
channel = 4
[EMAIL PROTECTED] ~]# cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[mobile]
exten = _906NXXX,1,Dial(Zap/1/${EXTEN:1})
exten = _906NXXX,2,Hungup()
[outbound]
exten = _9ZX.,1,Dial(Zap/1/${EXTEN:1})
exten = _9ZX.,2,Hangup()
[voicemail]
exten = 31,1,VoiceMailMain([EMAIL PROTECTED])
exten = 33,1,VoiceMailMain([EMAIL PROTECTED])
[konferencija]
exten = 40,1,Meetme(40,s)
exten = 40,2,Hangup()
[interno]
exten = 21,1,Dial(SIP/maja,20)
exten = 21,2,Hangup()
exten = 24,1,Dial(SIP/esad,20)
exten = 24,2,Hangup()
[local]
exten = 11,hint,SIP/cisco1
exten = 11,1,Dial(SIP/cisco1,20)
exten = 11,2,Hangup()
exten = 12,hint,Zap/3
exten = 12,1,Dial(Zap/3,20)
exten = 12,2,Hangup()
exten = 13,hint,SIP/sipura
exten = 13,1,Dial(SIP/sipura,20)
exten = 13,2,Hangup()
exten = 14,hint,SIP/goran
exten = 14,1,Dial(SIP/goran,20)
exten = 14,2,Hangup()
exten = 15,hint,SIP/bobana
exten = 15,1,Dial(SIP/bobana,20)
exten = 15,2,Hangup()
exten = 16,hint,SIP/miroslav
exten = 16,1,Dial(SIP/miroslav,20)
exten = 16,2,Hangup()
exten = 17,hint,SIP/pop
exten = 17,1,Dial(SIP/pop,20)
exten = 17,2,Hangup()
exten = 18,hint,SIP/zoran
exten = 18,1,Dial(SIP/zoran,20)
exten = 18,2,Hangup()
exten = 20,hint,SIP/dusan
exten = 20,1,Dial(SIP/dusan,20)
exten = 20,2,Hangup()
include = outbound
include = mobile
include = konferencija
include = voicemail
[incoming]
exten = 11,1,Dial(SIP/cisco1,20)
exten = 11,2,VoiceMail([EMAIL PROTECTED])
exten = 11,3,Playback(vm-goodbye)
exten = 11,4,Hangup()
exten = 11,102,VoiceMail([EMAIL PROTECTED])
exten = 11,103,Hangup()
exten = 12,1,Dial(Zap/3,20)
exten = 12,2,Playback(vm-goodbye)
exten = 12,3,Hangup()
exten = 12,102,Playback(tt-allbusy)
exten = 12,103,Hangup()
exten = 13,1,Dial(SIP/sipura,20)
exten = 13,2,VoiceMail([EMAIL PROTECTED])
exten = 13,3,Playback(vm-goodbye)
exten = 13,102,VoiceMail([EMAIL PROTECTED])
exten = 13,103,Hangup()
exten = 14,1,Dial(SIP/zoran,20)
exten = 14,2,VoiceMail([EMAIL PROTECTED])
exten = 14,3,Playback(vm-goodbye)
exten = 14,102,VoiceMail([EMAIL PROTECTED])
exten = 14,103,Hangup()
exten = 15,1,Dial(SIP/rzoran,20)
exten = 15,2,VoiceMail([EMAIL PROTECTED])
exten = 15,3,Playback(vm-goodbye)
exten = 15,102,VoiceMail([EMAIL PROTECTED])
exten = 15,103,Hangup()
exten = 17,1,Dial(SIP/pop,20)
exten = 17,2,VoiceMail([EMAIL PROTECTED])
exten = 17,3,Playback(vm-goodbye)
exten = 17,102,VoiceMail([EMAIL PROTECTED])
exten = 17,103,Hangup()
exten = 20,1,Dial(SIP/dusan,20)
exten = 20,2,VoiceMail([EMAIL PROTECTED])
exten = 20,3,Playback(vm-goodbye)
exten = 20,102,VoiceMail([EMAIL PROTECTED])
exten = 20,103,Hangup()
exten =
Re: [asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
My question is does ANYONE do ANY testing on these releases? It would seem that this bug is so paramount to the purpose of the code that had anyone taken a MINUTE to TEST, it would have been discovered IMMEDIATELY. sigh. Thanks, Steve Totaro On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sean Bright to Asterisk show details 4:47 PM (15 hours ago) There is a bug in 'make install' in Zaptel 1.4.10 that causes the devices to not be installed correctly. You can either install 1.4.9 or wait for 1.4.11 to be released. On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o. [EMAIL PROTECTED] wrote: Hi list! I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linux with installed digium packets 1. Asterisk 1.4.19 2. Zaptel 1.4.10 3. Libpri 1.4.3 My Digium hardware is [EMAIL PROTECTED] ~]# zaptel_hardware pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card The problem is the asterisk doesn't recognize the Zap channels at all. The error is No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) and there is the original output form Astersik console: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20) in new stack [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type registered for 'Zap' [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new stack == Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90' And everything was working quite fine when I was on asterisk 1.2.13, previously installed on this very same server, same Digium card etc. The configurations are totaly the same, also. What could be the resolution of this problem? ___ -- 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] Zap channels for HFC-S PCI card not responding
On Sun, Dec 30, 2007 at 04:48:39PM +0100, Jaap Winius wrote: Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? What do you mean by busy? What exactly do you see? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Zap channels for HFC-S PCI card not responding
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What do you mean by busy? What exactly do you see? This kind of thing: # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) Any attempts to call out result in the following CLI output: [Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION' [Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081ff9f8' not posted CLI zap restart: Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! This and more is from my previous message (sorry, that didn't just contain configuration information). Thanks, Jaap ___ --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] Zap channels: no sound with certain call paths
On Wed, Sep 12, 2007 at 09:08:23PM -0400, Christian Weeks wrote: Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house phones. Also, a Wildcard TDM400P, using the wctdm module with 1 FXO module, this supplies FXO to the upstream telco (a single line). The problem: Lately, and without any configuration changes, incoming calls that route through the Wildcard (from the telco) to the channel bank (well, a phone connected to the channel bank) have no voice in either direction. Obviously, this is rather frustrating. The same configuration has worked quite reliably for the past year or so, so I am reasonably confident that the problem isn't directly configuration related (though I have, since this started occuring tried various configs). The version where this started to occur (intermittently) was asterisk/zaptel in debian etch (the 1.2 branch). I have since upgraded to zaptel/asterisk from debian sid (the 1.4 branch) and the problems have gotten marginally worse. Stuff I have tried: 1. Zap-Zap (calling one channel bank extn from another) works fine. 2. Zap-anywhere (calling out from CB to telco through wildcard, or to SIP provider, or IAX provider) works fine. 3. telco-Zap (calling in from telco to CB line) fails: no voice. 4. SIP/IAX-Zap (calling in from a SIP client to CB line) works. Diagnostics examined: 1. ztmonitor any line -v shows expected signals, from the asterisk perspective. But e.g. in scenario 3 above, there is no received voice from the zap line. Which is consistent with the dialled CB line not being properly connected somehow. Oddities noticed: 1. Sometimes, when picking up a CB line, there is no dialtone. Only resolution has been to reset the computer. 2. There are several odd messages in the log files: (/var/log/syslog) [..snip..] Sep 12 17:52:04 phone kernel: Got pulse digit 36 on R1T1/0/3?? (note: lots of these, at least one per CB line, whenever we restart or reprobe the module) This means many close on-hook/of-hook events. Close enough to create 36 pulse dials. This is from zaptel.ko . [..snip..] Sep 12 17:53:29 phone asterisk[2638]: rc_avpair_new: unknown attribute 1490026597 (lots of these too, there seems to be a correlation between these messages and no voice routings) (/var/log/asterisk/messages (I have verbosity up nice and high)) Make sure you have debug enabled and logged if you have strange things in chan_zap and want to full understand them. What version of Asterisk is it? [Sep 12 20:35:20] WARNING[3174] chan_zap.c: Ring/Off-hook in strange state 6 on channel 25 (I've had this since I set the environment up. No one seems to be able to give a sane answer as to why). Finally, here's an interesting oddity. I can get the voice to come up, in certain circumstances, by doing the following: 1. Dial in from telco using cellphone. 2. Answer with CB Zap line. No voice. 3. Hang up the CB Zap line. 4. Re-open any Zap CB line, execute a dial that uses telco line. 5. The telco line picks up (to execute the dial); voice is now connected to the still waiting original call. Here's the log file: [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Starting simple switch on 'Zap/25-1' [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing [EMAIL PROTECTED]:1] Goto(Zap/25-1, incoming-home|s|1) in new stack [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto (incoming-home,s,1) [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing [EMAIL PROTECTED]:1] NoOp(Zap/25-1, Number) in new stack [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing [EMAIL PROTECTED]:2] Set(Zap/25-1, TRANSFER_CONTEXT=transfer) in new stack [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing [EMAIL PROTECTED]:3] GotoIfTime(Zap/25-1, 9:00-20:00|*|*|*?s-DAY|1) in new stack [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto (incoming-home,s-DAY,1) [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(Zap/25-1, Zap/1Zap/3Zap/2Zap/10Zap/5Zap/6SIP/cpw...) [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 1 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 3 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 2 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 10 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 5 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 6 [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called me [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/1-1 is ringing [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/3-1 is ringing [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/2-1 is ringing [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/10-1 is ringing [Sep 12
Re: [asterisk-users] Zap channels unavailable?
Hi, I was talking to a technican at our telco yesterday and he told me that this problem was most likely caused by our PBX sending channel identification Exclusive when we dial out. If there's a heavy load and someone is dialing in on the same time on the same channel that we try to dial out from - it causes a deadlock. He said some Cisco PBXs have the same problem. Now, I'm no asterisk expert and I don't quite understand what this means. I've emailed the list asking if this can be changed to Preferred or Negotiation as the technican told me to. But I got no response yet. I did however solve the problem by reversing the channels that we dial out from (so now it tries the last channel first and then backwards to the first). Since all of our incoming calls come from the first to the last this minimizes the risk of a collision of the incoming/outgoing calls. This is of cource no long-term solution but anyway. I need to know if it's possible to change channel identification (whatever that is) to preferred or negotiation. Regards, Jan Martin Smith wrote: Hello Jan, We have also been seeing this issue, and we are running Asterisk 1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI provider that a 3rd party vendor has applied firmware to some hardware along our path, and that it has an unfortunate bug of hanging B-channels in the PRI flags resetting state. We have been assured that the vendor has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the problem, and that it will go away soon. In the mean time, we've also had to restart Asterisk to free our B-channels for use, and any link-level event potentially re-hangs them. Keep us posted if you find out anything! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jan.sarin at securia.se Sent: Tuesday, July 17, 2007 9:44 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Zap channels unavailable? Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --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] Zap channels unavailable?
Hello Jan, We have also been seeing this issue, and we are running Asterisk 1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI provider that a 3rd party vendor has applied firmware to some hardware along our path, and that it has an unfortunate bug of hanging B-channels in the PRI flags resetting state. We have been assured that the vendor has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the problem, and that it will go away soon. In the mean time, we've also had to restart Asterisk to free our B-channels for use, and any link-level event potentially re-hangs them. Keep us posted if you find out anything! Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 17, 2007 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zap channels unavailable? Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --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] Zap channels unavailable?
Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, [EMAIL PROTECTED] wrote: Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --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] Zap channels unavailable?
Hi, No I havn't tried that. That entry wasn't even in there so I'll try it. I'll let you know if it helped. The odd thing is that this problem started yesterday. And our asterisk has been running for +1 year without these kind of problems. So either our telco has changed something OR it's because of the heavy load on the server (cpu running at max 20% with 40-50 simultaneous calls, so why would it be this?). Regards, Jan -- Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote: Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --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] Zap channels unavailable?
Okay, I've got an update on the resetinterval=never... same thing even though i added the line to zapata.conf and restarted the server. Now the load wasn't even high, maybe 6-7 calls. I think I just might call my telco, feels like it's their issue, but if anyone has any other suggestions let me know and I'll try them! Channel: 7 File Descriptor: 17 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 708307496 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook Regards, Jan -Ursprungligt meddelande- Från: Jan Sarin Skickat: den 17 juli 2007 16:57 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: SV: [asterisk-users] Zap channels unavailable? Hi, No I havn't tried that. That entry wasn't even in there so I'll try it. I'll let you know if it helped. The odd thing is that this problem started yesterday. And our asterisk has been running for +1 year without these kind of problems. So either our telco has changed something OR it's because of the heavy load on the server (cpu running at max 20% with 40-50 simultaneous calls, so why would it be this?). Regards, Jan -- Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote: Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --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] Zap channels staying offhook - restart required
Try setting AbsoluteTimeout() as the first parameter in your dialplan entry. Check it out on voip-info.org On 1/28/07, kjcsb [EMAIL PROTECTED] wrote: Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --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] Zap channels staying offhook - restart required
Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --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] Zap channels staying offhook - restart required
Just for giggles can you set an absolute timeout in the dialplan for all calls in and out of that span? On 1/25/07, kjcsb [EMAIL PROTECTED] wrote: I have a situation where the two Zap channels on a TDM400 are staying offhook after a random period of time; it is not (I believe) related to the FXO side not hanging up. Actually I suspect the server is overheating but I need to do more analysis. Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? Also suggestions on debugging this would be appreciated. Regards Cameron ___ --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] Zap channels ringing too loudly
On 5/31/06, Nick Burch [EMAIL PROTECTED] wrote: Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring someone else) Having had a look through the code, I think that Asterisk passes the responsibility for ringing the phones to Zaptel, which drives the astribank to make them ring. Is this correct? Despite looking through the zaptel source code, I couldn't find anywhere that screamed I'm the volume your phones ring at. Just a lot of scary numbers in zonedata.c, and cryptic comments in tone_zone.h Could someone suggest how I'd go about making the zap ring volume quieter? I could be way off here, but I thought FXS ringing was signaled only by a change in voltage on the pair, so I'm not sure how zaptel could instruct the hardware device to send a different voltage? I think its only capability with FXS is to fluctuate the voltage to support distinctive rings. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [Asterisk-Users] Zap Channels , for round-robin search and call
Why not just define a group and use :-
exten = _9X.,1,Dial(ZAP/g1/${EXTEN:1})
On Wed, 2006-05-31 at 13:08, John Joseph wrote:
Hi
I am using a 4FXO , TDM400P card
I am able to call outside , after modifiying
extensions.conf
with
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
using this , I can only dial through one of the
port , Actually I want to dial outside using round -
robin search
After reading the manuals , I have plans to
modified the above line as
exten =
_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})
Please let me know wheter the above line , is
correct to use
I think , it will dial any one of the four
channel which is available
Please give your comments on the putting
the line
exten =
_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})
Thanks
Joseph John
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Re: [Asterisk-Users] Zap Channels , for round-robin search and call
depending on your zapata.conf file, you should use
exten = _9X.,1,Dial(Zap/r1/${EXTEN:1})
The little 'r' means round robin, starting at the next highest channel than last time.
Have a look in extensions.conf from the samples for more options.
Make sure you have your 4 channels in one group (group=1).
K
On 5/31/06, John Joseph [EMAIL PROTECTED] wrote:
HiI am using a 4FXO , TDM400P cardI am able to call outside , after modifiyingextensions.conf
withexten = _9X.,1,Dial(ZAP/1/${EXTEN:1})using this , I can only dial through one of theport , Actually I want todial outside using round -robinsearchAfter reading the manuals , I have plans to
modified the above line asexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})Please let me know wheter the above line ,iscorrect to useI think , it will dial any one of the four
channel which is available Pleasegive your comments on theputtingthe lineexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1}) Thanks
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Re: [Asterisk-Users] Zap channels ringing too loudly
I could be way off here, but I thought FXS ringing was signaled only by a change in voltage on the pair, so I'm not sure how zaptel could instruct the hardware device to send a different voltage? I think its only capability with FXS is to fluctuate the voltage to support distinctive rings. There may be cadences to adjust. It depends too on the country. ___ --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] Zap channels ringing too loudly
Hi Nick, On Wed, May 31, 2006 at 12:58:55PM +0100, Nick Burch wrote: Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring someone else) Having had a look through the code, I think that Asterisk passes the responsibility for ringing the phones to Zaptel, which drives the astribank to make them ring. Is this correct? Anything that analog is probably the job of the digital-to-analog chip used. Not of the digital stream sent by Zaptel. Despite looking through the zaptel source code, I couldn't find anywhere that screamed I'm the volume your phones ring at. Just a lot of scary numbers in zonedata.c, and cryptic comments in tone_zone.h Could someone suggest how I'd go about making the zap ring volume quieter? I don't have the specs here. I'll just note that component used in the Astribank is quite similar to the one used by Digium in the TDM400P and TDM2400P . Grep for proslic. Note, however: $ cat /proc/xpp/XBUS-0/XPD-0/slics # Writing bad data into this file may damage your hardware! # Consult firmware docs first SLIC_REPLY: D reg_num=0x0, dataH=0x0 dataL=0x0 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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
Re: [Asterisk-Users] Zap channels ringing too loudly
On Wednesday 31 May 2006 07:58, Nick Burch wrote: I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring someone else) I suppose it is possible that the Astribank is ringing 'hot' but honestly almost every single phone today is electronically rung and not mechanically rung... I can't imagine significant difference in ring volume due to higher ring voltage on any modern phone. These aren't old carbon-granule phones with mechanical bell ringers, are they? -A. ___ --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] Zap channels ringing too loudly
This is entirely your phones.. not asterisk... that is how loud the phones are set to ring.. is there a ringer setting on them? On 5/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Hi Nick, On Wed, May 31, 2006 at 12:58:55PM +0100, Nick Burch wrote: Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring someone else) Having had a look through the code, I think that Asterisk passes the responsibility for ringing the phones to Zaptel, which drives the astribank to make them ring. Is this correct? Anything that analog is probably the job of the digital-to-analog chip used. Not of the digital stream sent by Zaptel. Despite looking through the zaptel source code, I couldn't find anywhere that screamed I'm the volume your phones ring at. Just a lot of scary numbers in zonedata.c, and cryptic comments in tone_zone.h Could someone suggest how I'd go about making the zap ring volume quieter? I don't have the specs here. I'll just note that component used in the Astribank is quite similar to the one used by Digium in the TDM400P and TDM2400P . Grep for proslic. Note, however: $ cat /proc/xpp/XBUS-0/XPD-0/slics # Writing bad data into this file may damage your hardware! # Consult firmware docs first SLIC_REPLY: D reg_num=0x0, dataH=0x0 dataL=0x0 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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
Re: [Asterisk-Users] Zap channels ringing too loudly
On 5/31/06, BJ Weschke [EMAIL PROTECTED] wrote: I could be way off here, but I thought FXS ringing was signaled only by a change in voltage on the pair, so I'm not sure how zaptel could instruct the hardware device to send a different voltage? I think its only capability with FXS is to fluctuate the voltage to support distinctive rings. Ring voltage in North America is supposed to be 90vAC at 20Hz. Assuming these are Western Electric 2500 sets or similar, then a less-wimpy ring voltage generator could very well make the phones ring louder. Fortunately, if these are Western Electric sets, then there should be a dial marked LOUD or HI on the underside of the phone. Turn that to the right to make the bells softer. -- Strom Carlson http://www.stromcarlson.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
Re: [Asterisk-Users] Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails)
On Tuesday 25 April 2006 05:50, Mike Garey wrote: When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and hangs up before the voicemail starts to record, an empty message will still be recorded and sent to the user. It sounds very much like you need disconnect supervision. http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision You'll need to see what your provider provides (if anything) and setup your zaptel.conf/zapata.conf accordingly. hads -- A fool and his money are soon using Windows. ___ --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] Zap channels - help
Hi Tzafrir, thank´s for your help.
My configurations:
#zaptel.confspan=1,2,0,cas,hdb3cas=1-31:1101loadzone=usdefaultzone=us
span=2,1,0,ccs,hdb3bchan=32-46dchan=47bchan=48-62loadzone=usdefaultzone=us
#zapata.conf[trunkgroup]
[channels]context=defaultswitchtype=euroisdnsignalling=pri_net;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yes
threewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=2callgroup=2immediate=nocallerid=asreceivedmusiconhold=default
group=2channel=32-46channel=48-62
2006/4/1, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Mar 31, 2006 at 10:36:02PM -0300, Josué Conti wrote: I am installing one asterisk, to establish connection with my PABX Siemens,
in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the commandexten = _ 19, 1,
dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial(SIP/8110-a729, zap/g2/1971411234|30) in new stack -- Called g2/1971411234 -- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1' == No one is available to answer at this timeHowever, when use the rule exten = _ 7xxx, 1, dial(zap/g2/${EXTEN}, 30) I obtain to call the branches pabx, normally.
-- Executing Dial(SIP/8110-71ee, zap/g2/7500|30) in new stack -- Called g2/7500 -- Zap/32-1 is ringing -- Zap/32-1 answered SIP/8110-71ee -- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1' == Spawn extension (default, 7500, 1) exited non-zero on 'SIP/8110-71ee' Somebody would have some idea to help in this case me? Greatings JosuéCould you please post your
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Re: [Asterisk-Users] Zap channels - help
On Fri, Mar 31, 2006 at 10:36:02PM -0300, Josué Conti wrote:
I am installing one asterisk, to establish connection with my PABX Siemens,
in ISDN, link went up normally, also I obtain to internally call the
branches the PABX, normally, but when I try to dial for the PSTN, through
pabx with the command exten = _ 19, 1,
dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error:
-- Executing Dial(SIP/8110-a729, zap/g2/1971411234|30) in new stack
-- Called g2/1971411234
-- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1'
== No one is available to answer at this
time
However, when use the rule exten = _ 7xxx, 1, dial(zap/g2/${EXTEN}, 30) I
obtain to call the branches pabx, normally.
-- Executing Dial(SIP/8110-71ee, zap/g2/7500|30) in new stack
-- Called g2/7500
-- Zap/32-1 is ringing
-- Zap/32-1 answered SIP/8110-71ee
-- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1'
== Spawn extension (default, 7500, 1) exited non-zero on 'SIP/8110-71ee'
Somebody would have some idea to help in this case me?
Greatings
Josué
Could you please post your zaptel.conf and zapata.conf ?
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Re: [Asterisk-Users] Zap Channels
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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Re: [Asterisk-Users] Zap Channels
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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--
.:FaberK:.
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Re: [Asterisk-Users] Zap Channels
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
What is PRITRUNK1? where is it defined?
How do you know something is wrong? Could you please paste the trace
from the logs/cli when verbosity is set to a high enough value? (e.g: 3)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] Zap Channels
PRITRUNK1 is defined into the extensions.conf globals:
--
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
PRITRUNK3=Zap/g3
--
Well I know what's happening, from my asterisk CDRs, and also from the PRI-CDRs.
We use a Teles.
2005/12/2, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
What is PRITRUNK1? where is it defined?
How do you know something is wrong? Could you please paste the trace
from the logs/cli when verbosity is set to a high enough value? (e.g: 3)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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--
.:FaberK:.
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Re: [Asterisk-Users] Zap Channels
voip-info is back up, at least for me ;-) Wiley Siler wrote: Is there a way to get what channels are not in use from the CLI? ZAP SHOW CHANNELS just lists the configed channels and ZAP SHOW CHANNEL N just returns OffHook as long as the phone is plugged in. This is using 2 TDM400 4 port FXO cards ustilizing 6 ports to a channel bank. The analog lines never show anything other than OffHook. sigh where is the Wiki when I need it... Thanks, Wiley ___ 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
RE: [Asterisk-Users] Zap channels busy. Have to soft hangup.
First, you may want to consider that you do not have enough zap channels. Can you tell us something about your system? How many lines do you have, and are you bridging incoming calls to an extension or flashing them through a pbx? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Sent: Thursday, April 21, 2005 12:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zap channels busy. Have to soft hangup. Hey everybody I am having really bad nightmares about this. Every day now our phone system has all of it's 4 zap channels full. I have to soft hangup zap/1-1 and zap/3-1. voip*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/4-1 (defaults1 ) Up Bridged Call Zap/3-1 Zap/3-1 (intern-post 9411 1 ) Up Dial Zap/g1/411|70 Zap/2-1 (defaults1 ) Up Bridged Call Zap/1-1 Zap/1-1 (intern-post 914105702452 1 ) Up Dial Zap/g1/14105702452|70 4 active channel(s) voip*CLI soft hangup zap/1-1 Requested Hangup on channel 'Zap/1-1' -- Hungup 'Zap/2-1' == Spawn extension (intern-post, 914105702452, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' voip*CLI soft hangup zap/1-2 zap/1-2 is not a known channel voip*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/4-1 (defaults1 ) Up Bridged Call Zap/3-1 Zap/3-1 (intern-post 9411 1 ) Up Dial Zap/g1/411|70 2 active channel(s) voip*CLI soft hangup Zap/4-1 Requested Hangup on channel 'Zap/4-1' -- Hungup 'Zap/4-1' == Spawn extension (intern-post, 9411, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' voip*CLI Also I have this weird thing where line Zap/3-1 rings and Zap/4-1 also picks up. Or if I dial an outside number if somebody dials in at the same time it happens that instead of dialing out I get the person who just dialed in. Rather confusing for both of us. Honestly I did dig around a lot and could not find any specifics about my issue. Any help highly appreciated. -- Thomas ___ 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
Re: [Asterisk-Users] Zap channels not hanging up...
Carlos Chavez wrote: I have 2 Asterisk servers that communicate with IAX2 between them and support multiple SIP clients each. Only one of them has Zap channels to the PSTN. I've been having problems because the Zap channels do not hang up when a sip client of the external server makes a call to the PSTN. SIP --- Asterisk IAX2 Asterisk --- Zap The local * server is using CVS-HEAD-03/08/05-16:08:10 and has 3 X100P cards. The remote server is using Stable 1.0.6. When I use a SIP phone on the local network the Zap channel hangs up properly, it only happens if the call comes from the remote server or it has happened a couple of times when I redirect my desk phone to my cell. Have a look at Bug 3813 and see if it fits with your experience. I suspect the echo cancellor and it would be interesting to see if the X100P has the same problem. I would encourage anyone who has been experiencing erratic or non - functioning busydetect to check it also. With the reliability I am now seeing with the latest FXO modules, I finally think I now have a production quality hardware setup. The inability to be able to fully adjust TX and RX gains on the FXO module to balance line loss to the PSTN is the only showstopper to me recommending the TDMXX solution for small setups. Regards, Richard ___ 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] Zap Channels Disappear???
On Thu, 24 Feb 2005 09:51:40 -0700, Chris Modesitt [EMAIL PROTECTED] wrote: Problem: Zap Channels Disappear @ random intervals. (Channels have disappeared on both gateways twice this week). My asterisk and libpri are built from the lastest 1.0 stable branch (2/14/05) my zaptel is built from the latest CVS head (2/18/05). Linux = RHEL 3.0 The only compile time change I made to zconfig.h is enable MMX for the drivers. Maybe this is a problem with enabling MMX in recent builds? OK, my problem isn't identical: instead of random it's consistantly on reboot; instead of RHEL I've got CentOS 3.4 (using [EMAIL PROTECTED]); and my error is can't load chan_zap.so I'm going to go rebuild zaptel without MMX enabled and see if I still have this problem. aaron.glenn ___ 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] Zap channels in AU hanging up on STD pips
On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
Try busydetect=no Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Monday, 31 January 2005 19:17 To: jurgen; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. I had the same problem (before I switched to Telstra ISDN). Increasing busycount to 8 fixed it. -Shaun Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
Guys, On a telsttra line you can have the STD pip's removed, I recently did this on a few on my lines. If you want the setting to ask telstra for, ask me off list and i'll try and find it. On Mon, 31 Jan 2005 16:51:38 +1100, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ 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
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
witht he bigt just ask provisioningto have NOPIPS set on the required lines. simple really On Mon, 31 Jan 2005 19:17:05 +1100, Howard Lowndes wrote: On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
Dear Howard, Which version of Asterisk are you running? On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41fdc41d213711706326924! ___ 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] Zap channels in AU hanging up on STD pips
On Tue, 2005-02-01 at 12:29, Stuart Elvish wrote: Dear Howard, Which version of Asterisk are you running? ext*CLI show version Asterisk CVS-HEAD-12/20/04-15:18:30 built by [EMAIL PROTECTED] on a i686 running Linux On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41fdc41d213711706326924! ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
Dear Howard, That is pretty much the latest version. In zapata.conf where you have callprogress=yes we have progzone=au. We also have default and load zones set to au in zaptel.conf. This should tell asterisk to look for Australian tones rather than the US ones which I assume it does by default. Hope this helps. Kind Regards Stuart On Tuesday, Feb 1, 2005, at 10:46 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-01 at 12:29, Stuart Elvish wrote: Dear Howard, Which version of Asterisk are you running? ext*CLI show version Asterisk CVS-HEAD-12/20/04-15:18:30 built by [EMAIL PROTECTED] on a i686 running Linux On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41feee08352724861317368! Stuart Elvish Business Development Manager TNet.com.au - Becoming Australia's Favourite Internet SERVICE Provider Mobile Telephone0433 133 601 (+61 433 133 601) Email Address [EMAIL PROTECTED] Direct Telephone08 9221 7874 (+61 8 9221 7874) Office Telephone1300 661 NET (1300 661 638) Direct Facsimile0433 133 598 (+61 433 133 598) Office Facsimile08 9221 3864 (+61 8 9221 3864) This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ 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] Zap channels in AU hanging up on STD pips
On Tue, 2005-02-01 at 14:27, Stuart Elvish wrote: Dear Howard, That is pretty much the latest version. In zapata.conf where you have callprogress=yes we have progzone=au. Ah ha, now that is a good point of which I was not aware. Many tks. We also have default and load zones set to au in zaptel.conf. Yes I have that set and similar in indications.conf. This should tell asterisk to look for Australian tones rather than the US ones which I assume it does by default. Hope this helps. Kind Regards Stuart On Tuesday, Feb 1, 2005, at 10:46 Australia/Perth, Howard Lowndes wrote: On Tue, 2005-02-01 at 12:29, Stuart Elvish wrote: Dear Howard, Which version of Asterisk are you running? ext*CLI show version Asterisk CVS-HEAD-12/20/04-15:18:30 built by [EMAIL PROTECTED] on a i686 running Linux On the earlier versions we had problems with the call progress detect disconnecting calls (not specifically related to STD pips but it may be of help), however with the newer version of Asterisk we don't seem to encounter this problem as they have included the tone definitions for Australia. Kind Regards Stuart On Monday, Jan 31, 2005, at 13:34 Australia/Perth, Howard Lowndes wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 !DSPAM:41feee08352724861317368! Stuart Elvish Business Development Manager TNet.com.au - Becoming Australia's Favourite Internet SERVICE Provider Mobile Telephone 0433 133 601 (+61 433 133 601) Email Address [EMAIL PROTECTED] Direct Telephone 08 9221 7874 (+61 8 9221 7874) Office Telephone 1300 661 NET (1300 661 638) Direct Facsimile 0433 133 598 (+61 433 133 598) Office Facsimile 08 9221 3864 (+61 8 9221 3864) This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Zap channels in AU hanging up on STD pips
Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ 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] Zap channels stuck in 'Rsrvd' state
Good Evening I found your post about this problem. Did you ever find a fix for it? I'm experiancing the same problem. Thanks. Quoting Steve Creel [EMAIL PROTECTED]: I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Zap Channels Hang
Mark, With CVS version are you using now?? is it working ok?? Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Mark Messmore, Technical Support, University Telcom Inc. Enviado el: Jueves 1 de Abril del 2004 10:38 Para: [EMAIL PROTECTED] Asunto: 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] 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 (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network9642555752 )Ring Dial Zap/g2/64255575 Zap/17-1 (defaults1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 9656990622 ) 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.700 (20040331) Information __ This message was checked by NOD32 Antivirus System. http://www.nod32.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
Re: [Asterisk-Users] Zap Channels Hang
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
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] 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 (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network9642555752 )Ring Dial Zap/g2/64255575 Zap/17-1 (defaults1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 9656990622 ) 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 [EMAIL PROTECTED] 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] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] 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 (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network9642555752 )Ring Dial Zap/g2/64255575 Zap/17-1 (defaults1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 9656990622 ) 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Zap Channels Hang
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Thursday, April 01, 2004 8:38 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] 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 (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network9642555752 )Ring Dial Zap/g2/64255575 Zap/17-1 (defaults1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 9656990622 ) 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman
Re: [Asterisk-Users] Zap Channels Hang
Hi, I have same problem with zap channels. I have E100P installed on my asterisk 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. Regards, Sergi - Original Message - From: Bisker, Scott (7805) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 01, 2004 7:17 PM Subject: RE: [Asterisk-Users] Zap Channels Hang 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Thursday, April 01, 2004 8:38 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luciano Ramos Sent: Thursday, April 01, 2004 6:24 AM To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Antonio Rabena Enviado el: Jueves 1 de Abril del 2004 05:02 Para: [EMAIL PROTECTED] 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 (ContextExtensionPri ) State Appl. Data Zap/31-1 (default9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network9682908972 )Ring Dial Zap/g2/68290897 Zap/30-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network9938415442 )Ring Dial Zap/g2/93841544 Zap/29-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network9966446872 )Ring Dial Zap/g2/96644687 Zap/28-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network9938716482 )Ring Dial Zap/g2/93871648 Zap/27-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network9686272242 )Ring Dial Zap/g2/68627224 Zap/26-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network9656277802 )Ring Dial Zap/g2/65627780 Zap/25-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/24-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network9631678382 )Ring Dial Zap/g2/63167838 Zap/23-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network9656990622 )Ring Dial Zap/g2/65699062 Zap/22-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network9656763882 )Ring Dial Zap/g2/65676388 Zap/21-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network9626622722 )Ring Dial Zap/g2/62662272 Zap/20-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 9642901182 )Ring Dial Zap/g2/64290118 Zap/19-1 (defaults1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network9656276402 )Ring Dial Zap/g2/65627640 Zap/18-1
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 my asterisk 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 [EMAIL PROTECTED] 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] Zap Channels Hang
our cvs is 02/25/04 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 my asterisk 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Zap Channels Hang
how about for the stable version? im using 0.7.2.. is there any known bugs in this release? should i upgrade to CVS? At 01:55 AM 4/2/2004, you wrote: On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote: Hi, I have same problem with zap channels. I have E100P installed on my asterisk 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 Regards, Antonio Rabena
RE: [Asterisk-Users] Zap channels stuck in 'Rsrvd' state
I've just started having the same problem here today. I did and upgrade over the weekend to Zaptel-0.9.0 and the release candidate for Asterisk-1.0 CVS 3/28/04. I have 6 Adtran 750 FXS_KS for all channels. 1 T-1PRI and one EM_W T-1. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Creel Sent: Monday, March 29, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zap channels stuck in 'Rsrvd' state I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
