Re: [asterisk-users] help!!! Internal extensions not connect
Hi Erik, thanks for your help, I found a solution, but this problem only happens when my server reboot. I put permissions in: chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi I think the problem is FreePBX not Asterisk. But when I reboot lost permissions againI don't understand. thanks for your help. -- Carem Gyssell Nieto Garcia -- _ -- 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] help!!! Internal extensions not connect
On Tue, Mar 02, 2010 at 08:28:52AM -0500, carem gyssell nieto wrote: Hi Erik, thanks for your help, I found a solution, but this problem only happens when my server reboot. I put permissions in: chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi hmod 777 normally means you did something wrong. I suspect the file had to be executable. In which case 755 (or: a+x) would have done the job just as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] help!!! Internal extensions not connect
You doesn't seem to have a proper context,extension,priority available for internal calls while you have one for outbound calls. To get more detailed help an even an answer you have to provide more info. The cli output while trying to setup an internal call will help. Erik On 2 mrt 2010, at 00:31, carem gyssell nieto wrote: I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN..my problem it only with internal extension. Please help...it's an Asterisk bug??. Thanks :S -- Carem Gyssell Nieto Garcia -- _ -- 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
Re: [asterisk-users] help!!! Internal extensions not connect
On Tue, Mar 2, 2010 at 1:51 AM, lesouvage i...@meetmecall.nl wrote: You doesn't seem to have a proper context,extension,priority available for internal calls while you have one for outbound calls. To get more detailed help an even an answer you have to provide more info. The cli output while trying to setup an internal call will help. Erik Or maybe you have an internal dialplan built into the phone which block your call to the local extensions. As Erik suggest you have to provide more details HTH, Ioan -- _ -- 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] Help with extensions - can't dial 700
Chris, Don't forget that a change in features.conf requires a restart of asterisk (or the modual features.c) - you can't get away with just a reload. On 5/17/05, Chris Mason [EMAIL PROTECTED] wrote: Thanks, I removed that and will test. I don't have an analog extension here, I am testing using SIP remotely, will have to go to the resort to test. Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] Help with extensions - can't dial 700
Just a shot in the dark: do you have features.conf with the following lines in it? Could be that asterisk's sample config file got in the way... Call Parking defaults to using extension 700: [general] parkext = 700 Mojo Chris Mason wrote: I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. From a SIP phone I can dial 700 and all other extensions. From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I dial 700. Extensions.conf [office] exten = 700,1,Dial(SIP/700,20) exten = 700,2,VoiceMail,u700 exten = 700,102,VoiceMail,b700 exten = 701,1,Dial(SIP/701,20) exten = 701,2,VoiceMail,u701 exten = 701,102,VoiceMail,b701 exten = 702,1,Dial(SIP/702,20) exten = 702,2,VoiceMail,u702 exten = 702,102,VoiceMail,b702 exten = 703,1,Dial(SIP/703,20) exten = 703,2,VoiceMail,u703 exten = 703,102,VoiceMail,b703 ... [zap-in] exten = s,1,NoOp(Starting Zap-in) include = office == zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes context=zap-in group=1 signalling=fxo_ks faxdetect=incoming channel = 1-12 Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] Help with extensions - can't dial 700
Thanks, I removed that and will test. I don't have an analog extension here, I am testing using SIP remotely, will have to go to the resort to test. Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] Help with extensions - can't dial 700
Check features.conf for parking extensions. There default is 700 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason Sent: Monday, May 16, 2005 8:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Help with extensions - can't dial 700 I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. From a SIP phone I can dial 700 and all other extensions. From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I dial 700. Extensions.conf [office] exten = 700,1,Dial(SIP/700,20) exten = 700,2,VoiceMail,u700 exten = 700,102,VoiceMail,b700 exten = 701,1,Dial(SIP/701,20) exten = 701,2,VoiceMail,u701 exten = 701,102,VoiceMail,b701 exten = 702,1,Dial(SIP/702,20) exten = 702,2,VoiceMail,u702 exten = 702,102,VoiceMail,b702 exten = 703,1,Dial(SIP/703,20) exten = 703,2,VoiceMail,u703 exten = 703,102,VoiceMail,b703 .. [zap-in] exten = s,1,NoOp(Starting Zap-in) include = office == zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes context=zap-in group=1 signalling=fxo_ks faxdetect=incoming channel = 1-12 Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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RE: [Asterisk-Users] Help with extensions
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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Re: [Asterisk-Users] Help with extensions
You read my mind. I was about to compose a response asking
why you'd strip digits when this message arrived.
Thanks.
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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Re: [Asterisk-Users] Help with extensions
Nope. Didn't work. When the call hits extensions.conf no matching
extension definition exists. I tried replacing line #2,
exten = s,2,VoiceMail2(${EXTEN}) with exten =
_u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
in the sip debug. It seems like this pattern matching should solve
the problem. It is unclear why the pattern is not matched. Is the
syntax correct?
-Steve
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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RE: [Asterisk-Users] Help with extensions
I have had the same issue when receiving a call from an IAX provider.
Here is what I did to solve it.
[from-proxy]
exten = .,1,Goto(voicemail-direct,s,1)
[voicemail-direct]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Not sure it does not pattern match like it should be the above worked
for me.
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Blair
Sent: Sunday, February 06, 2005 8:43 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help with extensions
Nope. Didn't work. When the call hits extensions.conf no
matching extension definition exists. I tried replacing line
#2, exten = s,2,VoiceMail2(${EXTEN}) with exten =
_u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
in the sip debug. It seems like this pattern matching should
solve the problem. It is unclear why the pattern is not
matched. Is the syntax correct?
-Steve
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b
already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls
arriving at Asterisk from my SIP proxy to be sent directly to
voicemail. I'd also like the appropriate greeting played
when the call
gets to voicemail.
My proxy prefixes the extension with a u or b based on SIP
response codes before relaying to Asterisk. So when the call
arrives it
is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded as
expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried the
following but Asterisk cannot find the extension with this set of
rules. Can someone explain how what I want can be accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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