Re: [Asterisk-Users] ignorepat not working - what might I have done?
On Fri, Aug 26, 2005 at 12:31:29PM -0500, Eric Wieling aka ManxPower wrote: ignorepat does not work for SIP since the dialtone is coming from the SIP device, not from Asterisk. You would need to set the phone up to continue dialtone after dialing 9. Not all phones support that. Hm. In this case, I suspect that the dial tone isn't coming from the phone... I have two possible places to route calls starting with 9, and only one of them is a SIP device... In fact, I just commented out the pattern that leads to the SIP device, so the only thing left is a transfer into a voice menu, and I still get no dial tone after the leading 9... Continuing thanks in advance for more ideas... :) -- Mason Loring Bliss [EMAIL PROTECTED]http://blisses.org/ I am a brother of jackals, and a companion of ostriches. (Job 30 : 29) ___ --Bandwidth and Colocation sponsored by Easynews.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] ignorepat not working - what might I have done?
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
Can someone suggest what I might have done wrong?
Thanks!
--
Mason Loring Bliss [EMAIL PROTECTED]http://blisses.org/
I am a brother of jackals, and a companion of ostriches. (Job 30 : 29)
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Re: [Asterisk-Users] ignorepat not working - what might I have done?
Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
ignorepat does not work for SIP since the dialtone is coming from the
SIP device, not from Asterisk.
You would need to set the phone up to continue dialtone after dialing 9.
Not all phones support that.
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Re: [Asterisk-Users] ignorepat not working - what might I have done?
ignorepat only works for analong phones connected to FXS modules.
Steve Maroney
On Fri, 26 Aug 2005, Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
Can someone suggest what I might have done wrong?
Thanks!
--
Mason Loring Bliss [EMAIL PROTECTED]http://blisses.org/
I am a brother of jackals, and a companion of ostriches. (Job 30 : 29)
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Re: [Asterisk-Users] ignorepat not working - what might I have done?
Steve Maroney wrote: ignorepat only works for analong phones connected to FXS modules. It also works for the IAXy and might work for MGCP and SCCP devices, since dialtone is controled by the PBX for those protocols. ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] ignorepat doesn't work
The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. On Apr 23, 2005, at 11:56 PM, Jaime Blanco wrote: Jerry, when you say digitmap, you mean in my extensions.conf file? Thanks. Jaime From: Jerry [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Date: Sat, 23 Apr 2005 19:44:20 -0500 Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 ___ 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] ignorepat doesn't work
Jerry wrote: The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. Grandstream BT phones don't have a digitmap feature. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] ignorepat doesn't work
Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348 Content-Transfer-Encoding: 7bit Content-Type: text/enriched; charset=US-ASCII thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: colorparam,,DEDE/paramI had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. /color thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. colorparam,,DEDE/param Not sure if this is a bug or a feature. /color probably intentional. colorparam,,DEDE/paramSo, try placing the ignorepat in your handset-contexts instead. /color Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348-- ___ 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] ignorepat doesn't work
ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348 Content-Transfer-Encoding: 7bit Content-Type: text/enriched; charset=US-ASCII thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: colorparam,,DEDE/paramI had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. /color thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. colorparam,,DEDE/param Not sure if this is a bug or a feature. /color probably intentional. colorparam,,DEDE/paramSo, try placing the ignorepat in your handset-contexts instead. /color Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk --Apple-Mail-18--1172348-- ___ 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] ignorepat doesn't work
Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] ignorepat doesn't work
Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] ignorepat doesn't work
Jerry, when you say digitmap, you mean in my extensions.conf file? Thanks. Jaime From: Jerry [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Date: Sat, 23 Apr 2005 19:44:20 -0500 Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 ___ 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] ignorepat changing the sound of dialtone
On Sun, Apr 10, 2005 at 09:47:36AM -0500, Andy Hamilton wrote: On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote: Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. This depends on what kind of phone you are using. Sorry - With standard POTS phones on a Digium TDM FXS interface. Thanks, 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] ignorepat changing the sound of dialtone
Howdie folks, Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. Thanks, 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
Re: [Asterisk-Users] ignorepat changing the sound of dialtone
This depends on what kind of phone you are using. With most (any?) SIP phones, nothing will be sent by the phone to the server until it actually dials (whereas Skinny phones sent out on/off hook and digits realtime). If you're using a Cisco phone with a sip image, my guess is that you can set something in dialplan.xml (or whichever file it is that the Ciscos look at to match numbers). That way, if you wanted someone to press 0 to get an outside line, the phone would see that an 0 was pressed and immediately dial. Obviously, you would catch this in Asterisk. Once *'s gotten the call from an extension that has dialed an 0, an RTP stream with the phone would commence and * would wait for the number that the party wishes to call (I believe there is a setting to change the tone... check your conf files or the wiki; I'm not sure where exactly it is. Once * matches a dialing pattern, it would dial out. Hope this helps. Andy Hamilton FWD 428726 On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote: Howdie folks, Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. Thanks, 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
[Asterisk-Users] Ignorepat with capi
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Re: [Asterisk-Users] Ignorepat with capi
massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Try:
exten = _0.,1,Dial(CAPI/xxx:b${exten:1})
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RE: [Asterisk-Users] Ignorepat with capi
Try this:
exten = _0.,1,Dial(CAPI/xxx:b${EXTEN:1})
The :1 tells it to use everything except the first digit.
Robert Jackson
-Original Message-
From: massimo [mailto:[EMAIL PROTECTED]
Sent: Friday, April 09, 2004 6:59 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ignorepat with capi
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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Re: [Asterisk-Users] Ignorepat with capi
On Fri, 2004-04-09 at 12:58, massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
I don't have CAPI but to get my analog to work I have
ignorepat = 9
exten = _9.,1,Dial(${DIALOUTANALOG}/${EXTEN:1})
--
Dave Cotton [EMAIL PROTECTED]
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[Asterisk-Users] ignorepat
Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone since I have placed ignorepat = 9 in the config file. Any ideas? rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat
Sip phones generate their own dialtone. The ignore pat option is meaningless with regard to SIP phones. I would check the Qrandstream's dialplan and see if you can program it to ignore the dialtone after a '9' is pressed. I had to do something similar for my Sipura SPA-2000. Steve. On Sunday 14 December 2003 12:18, Burak Balasaygun wrote: Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone since I have placed ignorepat = 9 in the config file. Any ideas? rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ignorepat doesn't work
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat = 9
exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten = _9[123456789]XXX,2,Congestion
this is properly included in the handsets' context but the dial tone
disappears after pressing 9.
am I missing something?
thanks in advance
regards
bk
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Re: [Asterisk-Users] ignorepat doesn't work
Hi bk,
On Wed, Jul 09, 2003 at 17:16:55 +0900, BK [address only for mailing lists] wrote:
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat = 9
exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten = _9[123456789]XXX,2,Congestion
this is properly included in the handsets' context but the dial tone
disappears after pressing 9.
am I missing something?
I had the same problem here and discovered that ignorepat only works if
it's placed in the actual incoming context of your channels and not if
it's included from another context. Not sure if this is a bug or a feature.
So, try placing the ignorepat in your handset-contexts instead.
Grtz,
Oliver
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Re: [Asterisk-Users] ignorepat doesn't work
thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk
