[asterisk-users] Strange Error
Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error
Please, show your dial plan and name your Asterisk version. You might be call the Dial application with incomplete arguments. jg Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error
On Thu, 3 Jul 2014, Andrew Colin wrote: Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ 1) Please choose a more meaningful subject. Lots of errors can be considered strange. (Note that actually, this is a warning, not an error.) 2) Please show a few more lines of console output (with verbose and debug set high) to give us some context. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error
Sound like chan_sip was not build. Just a guess: check that openssl-dev is available --- Dennis Guse On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin and...@vsave.co.za wrote: Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Error
Hi Guys, Anyone ever seen this before. on asterisk 1.8 if i set one of my pabx extensions to show private number and send a call over VoIP with g729 the call fails but with alaw it works. If i enable the callerid on g729 it also works see error below From: Anonymoussip:anonymous@anonymous.invalid;user=phone;tag=07d44838 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error
On 25/09/13 15:42, Andrew Colin wrote: Hi Guys, Anyone ever seen this before. on asterisk 1.8 if i set one of my pabx extensions to show private number and send a call over VoIP with g729 the call fails but with alaw it works. If i enable the callerid on g729 it also works see error below From: Anonymoussip:anonymous@anonymous.invalid;user=phone;tag=07d44838 That single line doesnt really help. You would need to give a full sip trace of a working withheld alaw can and a working and non working g729 call. Also the asterisk console output for the failed call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error -- ASterisk 1.6
We've been here, done this; This is a 1.6 NEW and Specific message to tell you that Asterisk can't start it's canary-monitor thread and that under certain conditions, you might be about to lock up. Look through the earlier posts in April. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Wednesday, April 28, 2010 2:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange Error -- ASterisk 1.6 All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks, Seann -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Error -- ASterisk 1.6
All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error -- ASterisk 1.6
Danny Nicholas wrote: We've been here, done this; This is a 1.6 NEW and Specific message to tell you that Asterisk can't start it's canary-monitor thread and that under certain conditions, you might be about to lock up. Look through the earlier posts in April. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Wednesday, April 28, 2010 2:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange Error -- ASterisk 1.6 All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks, Seann Danny, Thanks for that response, it gave me just enough to confirm my idea. I can't find the stuff in the earlier threads (yet) but as i have a lot to shuffle through, and see what else I can find from it. Once again, thank you. Regards, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange error, logger.c: No more room in scheduler...
I found no info about this strange error: logger.c: No more room in scheduler logger.c: Asked to delete sched id -1??? Only in verbose mode. Someone know how to solve this? Asterisk 1.2.13 with sangoma A104EC Hints? Thnks. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --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] Strange error, logger.c: No more room in scheduler...
Does this occur in the latest 1.2.17 release? On 4/10/07, Massimo Nuvoli [EMAIL PROTECTED] wrote: I found no info about this strange error: logger.c: No more room in scheduler logger.c: Asked to delete sched id -1??? Only in verbose mode. Someone know how to solve this? Asterisk 1.2.13 with sangoma A104EC Hints? Thnks. ___ --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] Strange error, logger.c: No more room in scheduler...
Sean Bright ha scritto: Does this occur in the latest 1.2.17 release? I dont know, this is a production system with 2 pri linked to telco and 2 pri linked to a pbx, i planned a large update but the release in use is the 1.2.13. And, also, i checked the changelog of the 1.2.17, and i found no reference to this kind of porblem solved. :-) Bye begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange error
Someone know why my asterisk gives me the following msgs? Thank you - Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Zhttp://82.51.224.34/http://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.51.224.34/ ___ --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] Strange Error when calling
Dear All, After doing the test everything went fine, Thanks Anthony for putting me on the right direction. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, I believe you where right the dial plan seems to have been missing the TRUNK= statement and I found one in the file extensions.conf but not the correct group I configured so I changed it and will test again. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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 -- Thx MAG -- Thx MAG ___ --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] Strange error
This post is from three days ago. Dont know if you found a solution or not. It sounds look to be your provider. What comes up in the CLI ? - Original Message - From: Crazy Boy To: asterisk-users@lists.digium.com Sent: Wednesday, July 26, 2006 3:39 AM Subject: [asterisk-users] Strange error Hi Friends,We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything. My intercom is also working fine always. What is this error? Please tell me the solution.Looking forward to your response.ThanksRegards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Strange Error when calling
Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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] Strange Error when calling
Dear Anthony, I believe you where right the dial plan seems to have been missing the TRUNK= statement and I found one in the file extensions.conf but not the correct group I configured so I changed it and will test again. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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 -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange error
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything. My intercom is also working fine always. What is this error? Please tell me the solution. Looking forward to your response. ThanksRegards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Error when calling
Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten => s,1,Answer exten => s,2,Dial(sip/100) exten => s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=>1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --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] Strange Error when calling
- Original Message -
From: Mohamed A. Gombolaty
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Wed, 26 Jul 2006 18:40:07 -0300
Subject: [asterisk-users] Strange Error when
calling
Dear All,
Greetings.
I have a strange problem in recieving calls on the pri the zaptel is
green and everything seems very well, but when a call comes I can see
the call along with the caller ID but then I get this strange message
which make the call hungup:
error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on
channel 0/18, span 1.
Usually on a PRI you will get the number that the person dialed, the DID or DDI
you might call it. In this case Asterisk will send it to an extension with that
number, not the 's' extension. Try adding an extension with the number that
does the same as your s extension to see if this is it. Or even:
exten = _X.,1,Noop(Hey they called ${EXTEN})
exten = _X.,n,Hangup
--
Thx
MAG
Joshua Colp
Digium
___
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Re: [asterisk-users] Strange Error when calling
This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten = s,1,Answer exten = s,2,Dial(sip/100) exten = s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --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
[Asterisk-Users] Strange error in debug file
Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I've got a whole load of them (328 in the last 5 minutes ...) Julian ___ 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] Strange error in debug file
Asterisk wrote: Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I'm guessing that happens when asterisk has hung up on some device but that device hasn't figured it out yet(therefore it's still trying to talk back to asterisk). Do you know if any calls had recently completed? It would be nice if the debug info gave more information about what device sent us this phantom data. I've got a whole load of them (328 in the last 5 minutes ...) I have seen this message, when dealing with my SPA2000. I am still testing VOIP providers for home use and have not turned my box up live, so I don't monitor the logs unless I am hacking on it and see something interesting go by. You could crank your verbosity up (3 minimum), turn on SIP and/or IAX debugs and wait a little more prepared to see if it happens again. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Strange error in debug file
Replies inline: Andrew Thompson wrote: Asterisk wrote: Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I'm guessing that happens when asterisk has hung up on some device but that device hasn't figured it out yet(therefore it's still trying to talk back to asterisk). Do you know if any calls had recently completed? Trouble is that we have 40ish agents making and receiving calls all the time, so it's difficult to pin this down to a particular device. I'm going to turn on SIP debugging for a little while as well. It would be nice if the debug info gave more information about what device sent us this phantom data. True, but all I've got is 20 or so of these lines in a row I've got a whole load of them (328 in the last 5 minutes ...) I have seen this message, when dealing with my SPA2000. I am still testing VOIP providers for home use and have not turned my box up live, so I don't monitor the logs unless I am hacking on it and see something interesting go by. You could crank your verbosity up (3 minimum), turn on SIP and/or IAX debugs and wait a little more prepared to see if it happens again. Verbosity is set at 15 :) Thanks. Julian. ___ 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] Strange error in debug file
Hmm, Part of the show channels gives me this: SIP/6028-b1ff (AgentQ s1 ) Up (None)(None) SIP/6011-da2f (AgentQ s1 ) Up (None)(None) SIP/6019-8fe7 (AgentQ s1 ) Up (None)(None) SIP/6019-f866 (AgentQ s1 ) Up (None)(None) SIP/6010-6bea (AgentQ s1 ) Up (None)(None) SIP/750-2657 (AgentQ s1 ) Up (None)(None) and sip show channels give me: pbx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 192.168.6.2016014000ccebd-f4 00101/00102 ulaw 192.168.6.23860192f49f93a014 00102/0 ulaw 192.168.6.2006022000cce63-c5 00101/00102 ulaw 192.168.6.2246006000cce63-d2 00101/00102 ulaw 192.168.6.18960280ed19a7f7c9 00102/0 ulaw 192.168.6.200602269d116d02f8 00102/00101 ulaw 192.168.6.18960280f1f96eb4c8 00102/00101 ulaw 192.168.6.20460113159fe13643 00102/00101 ulaw 192.168.6.23860191feb9e9b68e 00102/00103 ulaw 192.168.6.23860192ece4e3548f 00102/00101 ulaw 192.168.6.23160103f4f548f0bf 00102/00103 ulaw 192.168.6.244750 316b21476bc 00102/00103 ulaw 12 active SIP channel(s) Does this mean I have a problem with my SIP channels ? All SIP devices are Cisco 7940 running SIP 7.3 Julian Andrew Thompson wrote: Asterisk wrote: Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I'm guessing that happens when asterisk has hung up on some device but that device hasn't figured it out yet(therefore it's still trying to talk back to asterisk). Do you know if any calls had recently completed? It would be nice if the debug info gave more information about what device sent us this phantom data. I've got a whole load of them (328 in the last 5 minutes ...) I have seen this message, when dealing with my SPA2000. I am still testing VOIP providers for home use and have not turned my box up live, so I don't monitor the logs unless I am hacking on it and see something interesting go by. You could crank your verbosity up (3 minimum), turn on SIP and/or IAX debugs and wait a little more prepared to see if it happens again. ___ 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] Strange error
Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. Im using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. image002.jpg___ 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] Strange error
you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa [EMAIL PROTECTED] wrote: Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. I'm using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. ___ 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] Strange error
I think it is a CODEC change. For some reason I dont have chan_oh323.c in my channel directory or I would tell you for sure. It is not some much an error as a notification, if the calls still work. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: 12 November 2004 12:16 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Strange error Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. Im using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. inline: image001.jpg___ 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] Strange error
This is the version with whom i compile asterisk-oh323 channel successfully. I try the latest version of both asterisk and asterisk-oh323, but impossible to compile -Original Message- From: Paradise Dove [mailto:[EMAIL PROTECTED] Sent: vendredi 12 novembre 2004 18:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Strange error you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa [EMAIL PROTECTED] wrote: Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. I'm using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. ___ 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 ___ 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] strange error at extension.conf
hi,
i write this looking for free conference room, i
checl code and don´t see any error but die at priority 7 if room 1001 have users
in
exten =
_1NXXNXX,1,RouteCall(${EXTEN})exten =
_1NXXNXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13)exten =
_1NXXNXX,3,Setvar,var=0exten =
_1NXXNXX,4,MeetMeCount(1001|var)exten =
_1NXXNXX,5,GotoIf($[${var} =0]?7:6)exten =
_1NXXNXX,6,Meetme(1001|M)exten =
_1NXXMXX,7,MeetMeCount(1002|var)exten =
_1NXXNXX,8,GotoIf($[${var} =0]?10:9)exten =
_1NXXNXX,9,Meetme(1002|M)exten =
_1NXXMXX,10,MeetMeCount(1003|var)exten =
_1NXXNXX,11,GotoIf($[${var} =0]?4:12)exten =
_1NXXNXX,12,Meetme(1003|M)exten =
_1NXXNXX,13,Dial(${DESTINATION1})exten =
_1NXXNXX,114,Congestion
meetme.conf :
conf = 1001conf = 1002conf =
1003
in this log all working ok, no one at room
1001
-- Executing RouteCall("SIP/3056236725-7dc3",
"16058475739") in new stackApr 13 07:09:59 DEBUG[21520]: pbx.c:1088
pbx_substitute_variables_helper: _expression_ is '1' --
Executing GotoIf("SIP/3056236725-7dc3", "1?3:13") in new
stack -- Goto
(internal,16058475739,3) -- Executing
SetVar("SIP/3056236725-7dc3", "var=0") in new stack --
Executing MeetMeCount("SIP/3056236725-7dc3", "1001|var") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Parsing
'/etc/asterisk/meetme.conf': FoundApr 13 07:09:59 WARNING[21520]:
ast_expr.y:346 ast_yyerror: ast_yyerror(): syntax error: parse errorApr 13
07:09:59 DEBUG[21520]: pbx.c:1088 pbx_substitute_variables_helper: _expression_ is
'0' -- Executing GotoIf("SIP/3056236725-7dc3", "0?7:6") in
new stack -- Goto
(internal,16058475739,6) -- Executing
MeetMe("SIP/3056236725-7dc3", "1001|M") in new stack == Parsing
'/etc/asterisk/meetme.conf': == Parsing '/etc/asterisk/meetme.conf':
Found -- Created ZapTel conference 1023 for conference
'1001' -- Playing 'conf-onlyperson' (language 'en')Apr
13 07:10:03 DEBUG[21520]: app_meetme.c:379 conf_run: Placed channel
SIP/3056236725-7dc3 in ZAP conf 1023 -- Started music on
hold, class 'default', on SIP/3056236725-7dc3 -- Stopped
music on hold on SIP/3056236725-7dc3 == Spawn extension (internal,
16058475739, 6) exited non-zero on 'SIP/3056236725-7dc3'
in this log you can see the error, one user is at 1001
-- Executing RouteCall("SIP/3056236725-6771",
"16058475739") in new stackApr 13 07:17:04 DEBUG[23569]: pbx.c:1088
pbx_substitute_variables_helper: _expression_ is '1' --
Executing GotoIf("SIP/3056236725-6771", "1?3:13") in new
stack -- Goto
(internal,16058475739,3) -- Executing
SetVar("SIP/3056236725-6771", "var=0") in new stack --
Executing MeetMeCount("SIP/3056236725-6771", "1001|var") in new stackApr 13
07:17:04 WARNING[23569]: ast_expr.y:346 ast_yyerror: ast_yyerror(): syntax
error: parse errorApr 13 07:17:04 DEBUG[23569]: pbx.c:1088
pbx_substitute_variables_helper: _expression_ is '1' --
Executing GotoIf("SIP/3056236725-6771", "1?7:6") in new
stack -- Goto (internal,16058475739,7)
here stop and die ( exten
= _1NXXMXX,7,MeetMeCount(1002|var) )Apr 13 07:17:14
WARNING[23569]: pbx.c:1837 ast_pbx_run: Timeout, but no rule 't' in context
'internal'
Please, any one can help me???
thanks,
Carlos
[EMAIL PROTECTED]
