Re: [asterisk-users] invalid extension
you should check dialstatus and gotoif. if you use both in the proper way ( see the wiki) then you have the dialplan behaviour you are looking for. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 21:26 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ Administrator TOOTAI escribió: Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1); accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? Hi, The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. As you can see in your call, it won't work directly because asterisk by default will reject a call that doesn't match in the context or included contexts you defined for the user. Because the call is not accepted there's no need for a hangup (in a SIP environment). If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup exten = _XX.,1,Goto(s,1); accept exten LEN 1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] invalid extension
Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] invalid extension
Administrator TOOTAI escribió: Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN 1 alpha exten = _X.,1,Goto(s,1) ; accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? Hi, The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. As you can see in your call, it won't work directly because asterisk by default will reject a call that doesn't match in the context or included contexts you defined for the user. Because the call is not accepted there's no need for a hangup (in a SIP environment). If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup exten = _XX.,1,Goto(s,1) ; accept exten LEN 1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] invalid extension
Miguel Molina a écrit : [...] The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. Well, from reading voip.org it's not really clear than ... [...] Because the call is not accepted there's no need for a hangup (in a SIP environment). Well, I like when logs are clear ;-) and not have to guess :-) If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup Did it but get 2 hangup! First calling 2...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Goto(SIP/sip.tootai.net-084b1dc8, h,1) in new stack -- Goto (from-guest,h,1) -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' Second calling h...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' exten = _XX.,1,Goto(s,1) ; accept exten LEN 1 numeric Here your calling a three or more digits ;-) That will be enough to hangup what you want to, adjusting it to your needs. I will leave with this :-) Many thanks for the informations. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Invalid Extension
Do you have extension ontext 059*162*178*122*78600051 in your extensions.conf under the default context ? - Original Message - From: Philipp Kempgen philipp.kemp...@amooma.de To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, February 02, 2009 10:40 PM Subject: Re: [asterisk-users] Invalid Extension David @ULC schrieb: vicidialnow*CLI -- Executing AGI(SIP/66.54.140.46-b7800468, agi-VDAD_ALL_inbound.agi|CIDL OOKUPRC-LB-SALESLINE-936998-Closer-park--999-1-- ---TESTCAMP) in new stack Feb 2 14:53:09 NOTICE[18377]: chan_local.c:526 local_alloc: No such extension/c ontext 059*162*178*122*78600...@default creating local channel Feb 2 14:53:09 NOTICE[18377]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/059*162*178*122*78600...@default Ugh. When I call my DID, it get answered at my end but at other end , customer hears Its an INVALID extension and line get hang up. What could be the reason ? Could be a bug in ViciDial or in your specific setup. http://astguiclient.sourceforge.net/vicidial.html http://www.eflo.net/vicidial.php Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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
[asterisk-users] Invalid Extension
CLI Output : vicidialnow*CLI == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Ringing(SIP/66.54.140.46-b7800468, ) in new stack -- Executing Wait(SIP/66.54.140.46-b7800468, 1) in new stack -- Executing Answer(SIP/66.54.140.46-b7800468, ) in new stack -- Executing AGI(SIP/66.54.140.46-b7800468, agi-VDAD_ALL_inbound.agi|CIDL OOKUPRC-LB-SALESLINE-936998-Closer-park--999-1-- ---TESTCAMP) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Feb 2 14:53:09 NOTICE[18377]: chan_local.c:526 local_alloc: No such extension/c ontext 059*162*178*122*78600...@default creating local channel Feb 2 14:53:09 NOTICE[18377]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/059*162*178*122*78600...@default == Parsing '/etc/asterisk/manager.conf': Found When I call my DID, it get answered at my end but at other end , customer hears Its an INVALID extension and line get hang up. What could be the reason ? Need a copy of Extension.conf file ? ___ -- 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] Invalid Extension
David @ULC schrieb: vicidialnow*CLI -- Executing AGI(SIP/66.54.140.46-b7800468, agi-VDAD_ALL_inbound.agi|CIDL OOKUPRC-LB-SALESLINE-936998-Closer-park--999-1-- ---TESTCAMP) in new stack Feb 2 14:53:09 NOTICE[18377]: chan_local.c:526 local_alloc: No such extension/c ontext 059*162*178*122*78600...@default creating local channel Feb 2 14:53:09 NOTICE[18377]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/059*162*178*122*78600...@default Ugh. When I call my DID, it get answered at my end but at other end , customer hears Its an INVALID extension and line get hang up. What could be the reason ? Could be a bug in ViciDial or in your specific setup. http://astguiclient.sourceforge.net/vicidial.html http://www.eflo.net/vicidial.php Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] invalid extension dilemma
In the example below if I dial valid extension 1000, the Invalid context plays pbx-invalid as it is included with _7 context. Include voicemail in the main context. ___ 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] invalid extension dilemma
u can use this:
exten = i,1,Playback(invalid_selection)
exten = i,2,Goto(inbound_menu,_X.,1)
Bruno.
Joseph wrote:
Ho do you folks solve the problem with invalid extension when someone
dials a wrong number?
For example if somebody dial prefix _7 I want to allow tall free
numbers from that line but not a long distance. However, if somebody
dial
wrong number I want to play invalid extension instead of congestion.
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
[goto-dialout]
exten = _9.,1,SetMusicOnHold(loud)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9.,3,Hangup()
exten = _71800XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71866XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71877XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71888XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,1,SetMusicOnHold(loud)
exten = _7NXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,3,Hangup()
include = invalid
[invalid]
exten = _.,1,NoCDR()
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
exten = 1000,4,Hangup()
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] invalid extension dilemma
On Wed, 2005-08-03 at 07:52 +0200, Wilson Pickett wrote: In the example below if I dial valid extension 1000, the Invalid context plays pbx-invalid as it is included with _7 context. Include voicemail in the main context. Thanks, I new it must be something simple. Simply reposition the context voicemail before goto-dialout did the trick. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] invalid extension dilemma
Ho do you folks solve the problem with invalid extension when someone
dials a wrong number?
For example if somebody dial prefix _7 I want to allow tall free
numbers from that line but not a long distance. However, if somebody
dial
wrong number I want to play invalid extension instead of congestion.
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
[goto-dialout]
exten = _9.,1,SetMusicOnHold(loud)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9.,3,Hangup()
exten = _71800XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71866XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71877XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71888XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,1,SetMusicOnHold(loud)
exten = _7NXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,3,Hangup()
include = invalid
[invalid]
exten = _.,1,NoCDR()
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
exten = 1000,4,Hangup()
--
#Joseph
--
#Joseph
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[Asterisk-Users] INVALID Extension
I have a Cisco 7960G hooked up to a VPN connection tied into an Asterisk box The problem I have is after a certain amount of time, when you try to contact that extension a message comes up I'm sorry, that's an invalid extension. It works sometimes, and then it stops. I changed the Registration expiration time from 3600 seconds to 60 seconds, it helped but it's still not resolved. - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Invalid extension handling
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten = i,1,Playback,invalid
exten = i,n,Goto(incoming,_NXXNXX,1)
;
[incoming]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Background(welcome); play welcome msg
ask for extension
exten = _NXXNXX,n,WaitExten(5) ; Wait for extension
This works fine, however, there is one special case that I would like to
handle differently. If the caller inadvertently presses the # key
following the extension, I would like to discard the # and then send the
call back onto the stack. I know how to strip the #, but I can't find
another command like WaitExten that will reprocess the call as new.
Any ideas are appreciated.
Thanks,
Adam
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RE: [Asterisk-Users] Invalid extension handling
Use Gotoif instead of Goto. Check Gotoif usage.
This will give you enough features to fork the calls after the extension
is re-entered
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, April 14, 2005 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Invalid extension handling
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten = i,1,Playback,invalid
exten = i,n,Goto(incoming,_NXXNXX,1) ;
[incoming]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Background(welcome); play welcome msg
ask for extension
exten = _NXXNXX,n,WaitExten(5) ; Wait for extension
This works fine, however, there is one special case that I would like to
handle differently. If the caller inadvertently presses the # key
following the extension, I would like to discard the # and then send the
call back onto the stack. I know how to strip the #, but I can't find
another command like WaitExten that will reprocess the call as new.
Any ideas are appreciated.
Thanks,
Adam
The contents of this email message and any attachments are confidential
and are intended solely for addressee. The information may also be
legally privileged. This transmission is sent in trust, for the sole
purpose of delivery to the intended recipient. If you have received this
transmission in error, any use, reproduction or dissemination of this
transmission is strictly prohibited. If you are not the intended
recipient, please immediately notify the sender by reply email and
delete this message and its attachments, if any.
NOTICE: If received in error, please destroy and notify sender. Sender does
not waive confidentiality or privilege, and use is prohibited.
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Re: [Asterisk-Users] Invalid extension handling
Adam Robins wrote:
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten = i,1,Playback,invalid
exten = i,n,Goto(incoming,_NXXNXX,1)
;
[incoming]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Background(welcome); play welcome msg
ask for extension
exten = _NXXNXX,n,WaitExten(5) ; Wait for extension
This works fine, however, there is one special case that I would like to
handle differently. If the caller inadvertently presses the # key
following the extension, I would like to discard the # and then send the
call back onto the stack. I know how to strip the #, but I can't find
another command like WaitExten that will reprocess the call as new.
Use Goto. Since you have a pattern of _2XXX if they dial 2XXX# then
Asteirsk will process the call as 2XXX and just discard the # since
it's not listening for DTMF since it's already hit the Dial.
Same for _NXXNXX.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] invalid extension (need help)
On April 13, 2005 12:35 am, amna saleem wrote: I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup You've already answered the call; no need to answer again, although it won't hurt. Make sure that these lines are either in the same context that your call is executing within, or that it is included in that context. -A. ___ 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] invalid extension(need help)
hi! I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? Thanx ___ 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] invalid extension (need help)
hi! I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? Thanx ___ 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] invalid extension (need help)
amna saleem wrote: hi! I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? if you've already answered the call earlier, you don't need to Answer it again in the invalid context. flynn ___ 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] Invalid Extension
I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? I had to change all occurance of s to 533990 in order for this to work. 533990 is my FWD #. May I ask how can I genearlize this using s? Regards, Norman Zhang [inbound-sip] exten = 533990,1,Answer exten = s,2,ResponseTimeout(5) exten = s,3,Background(mymenu) exten = t,1,Goto(s,2) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Goto(local,101,1) exten = 138,1,Goto(local,138,1) When you register with FWD, you used something like: register=userid:[EMAIL PROTECTED]/533990 where you've included /533990 at the end. That is telling FWD what exten number to send to your * box when receiving a call. Remove that and the 's' extension will work just fine. The 's' extension is a special start case that does not expect any digits to be passed to it from FWD in this case. So, you can use either approach in the dialplan, but you need to be consistent throughout. ___ 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] Invalid Extension
Hi,
I can dial-in and here the prompt, but whenever I select 101, I get
invalid extension. May I ask, is this the right way of answering
incoming calls?
Regards,
Norman Zhang
[inbound-sip]
exten = 533990,1,Answer
exten = s,2,ResponseTimeout(5)
exten = s,3,Background(mymenu)
exten = t,1,Goto(s,2)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(s,2)
exten = 101,1,Goto(local,101,1)
exten = 138,1,Goto(local,138,1)
;exten = 533990,1,Goto(local,101,1)
; Internal Extensions
[local]
exten = 101,1,Dial(${MAINPHONE},20,Tt)
exten = 101,2,Voicemail(u101)
exten = 101,3,Hangup
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Re: [Asterisk-Users] Invalid Extension
I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? I had to change all occurance of s to 533990 in order for this to work. 533990 is my FWD #. May I ask how can I genearlize this using s? Regards, Norman Zhang [inbound-sip] exten = 533990,1,Answer exten = s,2,ResponseTimeout(5) exten = s,3,Background(mymenu) exten = t,1,Goto(s,2) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Goto(local,101,1) exten = 138,1,Goto(local,138,1) ___ 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] Invalid Extension
Norman Zhang wrote: Hi, I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? Regards, Norman Zhang [inbound-sip] exten = 533990,1,Answer exten = s,2,ResponseTimeout(5) exten = s,3,Background(mymenu) Bearing in mind that the extensions are = extension, priority, something to do, you seem to be missing s,1... -- Cheers, Matt Riddell ___ Daily Asterisk News: http://www.sineapps.com/news.php for html http://www.sineapps.com/rssfeed.php for rss ___ 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] Invalid Extension
I can dial-in and here the prompt, but whenever I select 101, I get invalid extension. May I ask, is this the right way of answering incoming calls? [inbound-sip] exten = 533990,1,Answer exten = s,2,ResponseTimeout(5) exten = s,3,Background(mymenu) Bearing in mind that the extensions are = extension, priority, something to do, you seem to be missing s,1... I need to replace all s with 533990, so * would answer to incoming calls. If I use s, I get error 404. Am I missing something? Regards, Norman Zhang ___ 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] invalid extension - missing the original ${EXTEN} value
How I can retrieve the original ${EXTEN} value when falling into the
exten = i,1,whatever context??
I'm trying to implement this extension rule:
exten = i,1,NoOp(${EXTEN})
exten = i,2,Wait(1)
exten = i,3,Playback(vm-extension)
exten = i,4,SayDigits(${EXTEN})
exten = i,5,Playback(is-invalid)
exten = i,6,Playback(pls-try-again)
exten = i,7,Goto(s,12)
But the ${EXTEN} in this example is i:
-- Invalid extension '4' in context 'callerid0800' on SIP/1018-e5b5
== CDR updated on SIP/1018-e5b5
-- Executing NoOp(SIP/1018-e5b5, i) in new stack
TIA, hermann
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Re: [Asterisk-Users] invalid extension - missing the original ${EXTEN} value
Hermann Wecke wrote:
How I can retrieve the original ${EXTEN} value when falling into the
exten = i,1,whatever context??
Try using ${INVALID_EXTEN}
--
Andres
Network Admin
http://www.telesip.net
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