Re: [asterisk-users] T1 problem (call using a .call file)
Thanks for your help Don’t really know the answer, but these are “givens”: > >1. your phone is (most likely) in the same area code as the asterisk >installation > > My phone has a different area code than the asterisk installation. The asterisk box is in FL and I can call a number in MN but not the 201 or many others > >1. >2. NY is most likely not in the same area code. > > I agree but I could call a MN cell phone for example which works all the time > >1. >2. Even though the T1 is a dedicated digital service, the code that >handles all of this is/was written to process calls from analog sources for >backwards compatibility and therefore would have the timing issue handlers >in place even though they don’t apply. > > > > My research revealed that you might use an exception to stop this, but I > didn’t really find a good example. You could check viop-info.org or > whirlpool to see what they say. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno > *Sent:* Friday, March 20, 2009 9:39 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) > > > > I still find it weird as even if it is a switch timing problem. Because > when is it calling my phone *all the time *and that other area code it *never > *calls it. Does that mean asterisk always complete my number in a certain > time frame, and the other number no? Also I get the progress code 127 > exactly after i move my call file to the outgoing folder, there is no delay, > I get it tthe same time I move the move. > > > > And also why the call goes through when I put SIP/whatever in the > callerid? Does that mean asterisk get to complete the call in the time > frame? > > On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas wrote: > > You can also do a set variable in the call file. I don’t really know how > to do that, but you can probably find the command and syntax on > voip-info.org. > > The reason it works on certain numbers has to do with switch timing. If * > can complete the call within a certain time frame, all is well. If not, the > 127 thing will bite you. > > You would think we were past that type of thing, but I suppose not. > > > > Another thing you might try is changing the 60 to 90 or so on your original > call file. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno > *Sent:* Thursday, March 19, 2009 4:42 PM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) > > > > I dont want to change it within my extensions.conf, because I have many > dids, and change them on the fly according to the call i am making. I have > a web interface where I fill a form that gets the number I am calling, the > caller id and context to go etc... > > > > I dont want to keep editing extensions.conf and reload, I want to do it > directly in the call file. > > > > What I dont understand is WHY it works on certain numbers and not all. > That is a problem, it is not normal. > > > > > > On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas wrote: > > GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the > trick > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle > Sent: Thursday, March 19, 2009 3:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] T1 problem (call using a .call file) > > Pascal Bruno wrote: > > Also very strange, when in my call file I change the callerid line to > > SIP/whatever like Danny said, the call go through, but I dont want > > that, because when I do so, it is displaying the main number on my T1 > > account as caller id and I dont want that, I want to display one of my > > other DID as callerid. > > > Then change your caller-id within your dialplan, not the callfile. > > Doug > > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or up
Re: [asterisk-users] T1 problem (call using a .call file)
Don't really know the answer, but these are "givens": 1. your phone is (most likely) in the same area code as the asterisk installation 2. NY is most likely not in the same area code. 3. Even though the T1 is a dedicated digital service, the code that handles all of this is/was written to process calls from analog sources for backwards compatibility and therefore would have the timing issue handlers in place even though they don't apply. My research revealed that you might use an exception to stop this, but I didn't really find a good example. You could check viop-info.org or whirlpool to see what they say. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Friday, March 20, 2009 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) I still find it weird as even if it is a switch timing problem. Because when is it calling my phone all the time and that other area code it never calls it. Does that mean asterisk always complete my number in a certain time frame, and the other number no? Also I get the progress code 127 exactly after i move my call file to the outgoing folder, there is no delay, I get it tthe same time I move the move. And also why the call goes through when I put SIP/whatever in the callerid? Does that mean asterisk get to complete the call in the time frame? On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas wrote: You can also do a set variable in the call file. I don't really know how to do that, but you can probably find the command and syntax on voip-info.org <http://voip-info.org/> . The reason it works on certain numbers has to do with switch timing. If * can complete the call within a certain time frame, all is well. If not, the 127 thing will bite you. You would think we were past that type of thing, but I suppose not. Another thing you might try is changing the 60 to 90 or so on your original call file. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Thursday, March 19, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas wrote: GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: > Also very strange, when in my call file I change the callerid line to > SIP/whatever like Danny said, the call go through, but I dont want > that, because when I do so, it is displaying the main number on my T1 > account as caller id and I dont want that, I want to display one of my > other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com <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 <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 <http://www.api-digital.com/> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
I still find it weird as even if it is a switch timing problem. Because when is it calling my phone *all the time *and that other area code it *never *calls it. Does that mean asterisk always complete my number in a certain time frame, and the other number no? Also I get the progress code 127 exactly after i move my call file to the outgoing folder, there is no delay, I get it tthe same time I move the move. And also why the call goes through when I put SIP/whatever in the callerid? Does that mean asterisk get to complete the call in the time frame? On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas wrote: > You can also do a set variable in the call file. I don’t really know how > to do that, but you can probably find the command and syntax on > voip-info.org. > > The reason it works on certain numbers has to do with switch timing. If * > can complete the call within a certain time frame, all is well. If not, the > 127 thing will bite you. > > You would think we were past that type of thing, but I suppose not. > > > > Another thing you might try is changing the 60 to 90 or so on your original > call file. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno > *Sent:* Thursday, March 19, 2009 4:42 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] T1 problem (call using a .call file) > > > > I dont want to change it within my extensions.conf, because I have many > dids, and change them on the fly according to the call i am making. I have > a web interface where I fill a form that gets the number I am calling, the > caller id and context to go etc... > > > > I dont want to keep editing extensions.conf and reload, I want to do it > directly in the call file. > > > > What I dont understand is WHY it works on certain numbers and not all. > That is a problem, it is not normal. > > > > > > On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas wrote: > > GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the > trick > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle > Sent: Thursday, March 19, 2009 3:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] T1 problem (call using a .call file) > > Pascal Bruno wrote: > > Also very strange, when in my call file I change the callerid line to > > SIP/whatever like Danny said, the call go through, but I dont want > > that, because when I do so, it is displaying the main number on my T1 > > account as caller id and I dont want that, I want to display one of my > > other DID as callerid. > > > Then change your caller-id within your dialplan, not the callfile. > > Doug > > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___ > -- 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 > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
You can also do a set variable in the call file. I don't really know how to do that, but you can probably find the command and syntax on voip-info.org. The reason it works on certain numbers has to do with switch timing. If * can complete the call within a certain time frame, all is well. If not, the 127 thing will bite you. You would think we were past that type of thing, but I suppose not. Another thing you might try is changing the 60 to 90 or so on your original call file. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Thursday, March 19, 2009 4:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas wrote: GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: > Also very strange, when in my call file I change the callerid line to > SIP/whatever like Danny said, the call go through, but I dont want > that, because when I do so, it is displaying the main number on my T1 > account as caller id and I dont want that, I want to display one of my > other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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 ___ -- 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] T1 problem (call using a .call file)
I dont want to change it within my extensions.conf, because I have many dids, and change them on the fly according to the call i am making. I have a web interface where I fill a form that gets the number I am calling, the caller id and context to go etc... I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file. What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal. On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas wrote: > GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the > trick > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle > Sent: Thursday, March 19, 2009 3:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] T1 problem (call using a .call file) > > Pascal Bruno wrote: > > Also very strange, when in my call file I change the callerid line to > > SIP/whatever like Danny said, the call go through, but I dont want > > that, because when I do so, it is displaying the main number on my T1 > > account as caller id and I dont want that, I want to display one of my > > other DID as callerid. > > > Then change your caller-id within your dialplan, not the callfile. > > Doug > > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___ > -- 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 > ___ -- 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] T1 problem (call using a .call file)
GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 19, 2009 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) Pascal Bruno wrote: > Also very strange, when in my call file I change the callerid line to > SIP/whatever like Danny said, the call go through, but I dont want > that, because when I do so, it is displaying the main number on my T1 > account as caller id and I dont want that, I want to display one of my > other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
Pascal Bruno wrote: > Also very strange, when in my call file I change the callerid line to > SIP/whatever like Danny said, the call go through, but I dont want > that, because when I do so, it is displaying the main number on my T1 > account as caller id and I dont want that, I want to display one of my > other DID as callerid. Then change your caller-id within your dialplan, not the callfile. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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] T1 problem (call using a .call file)
2608 __ast_read: DTMF end accepted
> without begin '3' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '3' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
> received on DAHDI/32-1, duration 0 ms
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
> without begin '1' on DAHDI/32-1
> [Mar 19 16:23:54] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
> passthrough '1' on DAHDI/32-1
> -- Executing [1201...@boxout:1] Dial("DAHDI/32-1",
> "DAHDI/g1/1201XXX") in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/1201XXX
> -- DAHDI/1-1 is proceeding passing it to DAHDI/32-1
> -- DAHDI/1-1 is making progress passing it to DAHDI/32-1
> -- DAHDI/1-1 is ringing
> -- DAHDI/1-1 answered DAHDI/32-1
> -- Native bridging DAHDI/32-1 and DAHDI/1-1
> -- Hungup 'DAHDI/1-1'
>
>
> Call is fine with the phone, but does not go through with .call file
>
>
>
>
>
>
> On Thu, Mar 19, 2009 at 11:47 AM, Danny Nicholas wrote:
>
>> Try this call file – replace XXX with your number and YYY with a valid
>> SIP exten on your system
>>
>>
>>
>> Channel: DAHDI/g1/1XX
>> Callerid: SIP/YYY
>>
>> MaxRetries: 1
>> RetryTime: 5
>> WaitTime: 60
>>
>>
>> --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
>> *Sent:* Thursday, March 19, 2009 9:22 AM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>>
>>
>>
>> Here is what my extensions.conf file has:
>>
>>
>>
>> exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
>> exten => _NXXNXX,n,Hangup()
>>
>>
>>
>> exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
>> exten => _1NXXNXX,n,Hangup()
>>
>>
>>
>> Using the phone, I can dial any numbers succesfully.
>>
>>
>>
>> And here is my call file:
>>
>>
>>
>> Channel: DAHDI/g1/1XX
>> Callerid: XX
>> MaxRetries: 1
>> RetryTime: 5
>> WaitTime: 60
>> Context: test
>> Extension: s
>> Priority: 1
>>
>>
>>
>> with the call file I can dial my cellphone which begin with 754XXX
>>
>> but when I call my friend's cellphone from new york which is 201XXX i
>> get progress code 127 as follows
>>
>>
>>
>> -- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 127 received
>>
>>
>>
>> I tried with the prefix 1 and without the prefix 1 it is always the same
>> thing, but with the handset I dial my phone and my friend's phone
>> succesfully with and without the 1
>>
>>
>>
>>
>>
>>
>>
>> On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas
>> wrote:
>>
>> Please paste the call file content (with the number ’ed of course) and
>> the Dial section from extensions.conf.
>>
>>
>> --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
>> *Sent:* Wednesday, March 18, 2009 6:24 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>>
>>
>>
>> This has to be a bug, because I dont know what else to try here
>>
>>
>>
>>
>>
>> On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno
>> wrote:
>>
>> Nope, I always dial 1 + 10 digits for all my numbers. It works on all
>> numbers when I am using my phone (Analogue or IP) but when I do it using a
>> .call file it does not work on some numbers mostly. That is the weirdest
>> thing I have ever seen. I tried different codecs in the call file, I still
>> get the PROGRESS with cause code 127
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg
>> wrote:
>>
>> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote:
>> > I have a weird problem with call using my T1 card. I can make calls
>
Re: [asterisk-users] T1 problem (call using a .call file)
DAHDI/1-1 is proceeding passing it to DAHDI/32-1
-- DAHDI/1-1 is making progress passing it to DAHDI/32-1
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered DAHDI/32-1
-- Native bridging DAHDI/32-1 and DAHDI/1-1
-- Hungup 'DAHDI/1-1'
Call is fine with the phone, but does not go through with .call file
On Thu, Mar 19, 2009 at 11:47 AM, Danny Nicholas wrote:
> Try this call file – replace XXX with your number and YYY with a valid
> SIP exten on your system
>
>
>
> Channel: DAHDI/g1/1XX
> Callerid: SIP/YYY
>
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Thursday, March 19, 2009 9:22 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> Here is what my extensions.conf file has:
>
>
>
> exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _NXXNXX,n,Hangup()
>
>
>
> exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _1NXXNXX,n,Hangup()
>
>
>
> Using the phone, I can dial any numbers succesfully.
>
>
>
> And here is my call file:
>
>
>
> Channel: DAHDI/g1/1XX
> Callerid: XX
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
> Context: test
> Extension: s
> Priority: 1
>
>
>
> with the call file I can dial my cellphone which begin with 754XXX
>
> but when I call my friend's cellphone from new york which is 201XXX i
> get progress code 127 as follows
>
>
>
> -- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
> -- Requested transfer capability: 0x00 - SPEECH
> -- PROGRESS with cause code 127 received
>
>
>
> I tried with the prefix 1 and without the prefix 1 it is always the same
> thing, but with the handset I dial my phone and my friend's phone
> succesfully with and without the 1
>
>
>
>
>
>
>
> On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas wrote:
>
> Please paste the call file content (with the number ’ed of course) and
> the Dial section from extensions.conf.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Wednesday, March 18, 2009 6:24 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> This has to be a bug, because I dont know what else to try here
>
>
>
>
>
> On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno wrote:
>
> Nope, I always dial 1 + 10 digits for all my numbers. It works on all
> numbers when I am using my phone (Analogue or IP) but when I do it using a
> .call file it does not work on some numbers mostly. That is the weirdest
> thing I have ever seen. I tried different codecs in the call file, I still
> get the PROGRESS with cause code 127
>
>
>
>
>
>
>
>
>
> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg
> wrote:
>
> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote:
> > I have a weird problem with call using my T1 card. I can make calls fine
> > using my analog and IP phones, but when I try to initiate a call using a
> > .call file, I get the following error
> > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- PROGRESS with cause code 127 received
> > it happens on certain numbers I dial, but if I dial that same number with
> an
> > ip or analog phone that use the T1 channel, the call is going through
> > normally.
> > Anybody knows why?
>
> Are you doing anything silly with prefixing or short-circuit dialing?
>
> in other words..
>
> You dial 8 for an outside line, then 1+10 digits
> and you're forgetting to do that for some numbers?
>
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>
>
>
>
>
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>
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Re: [asterisk-users] T1 problem (call using a .call file)
Try this call file - replace XXX with your number and YYY with a valid SIP
exten on your system
Channel: DAHDI/g1/1XX
Callerid: SIP/YYY
MaxRetries: 1
RetryTime: 5
WaitTime: 60
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Thursday, March 19, 2009 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)
Here is what my extensions.conf file has:
exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _NXXNXX,n,Hangup()
exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1NXXNXX,n,Hangup()
Using the phone, I can dial any numbers succesfully.
And here is my call file:
Channel: DAHDI/g1/1XX
Callerid: XX
MaxRetries: 1
RetryTime: 5
WaitTime: 60
Context: test
Extension: s
Priority: 1
with the call file I can dial my cellphone which begin with 754XXX
but when I call my friend's cellphone from new york which is 201XXX i
get progress code 127 as follows
-- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received
I tried with the prefix 1 and without the prefix 1 it is always the same
thing, but with the handset I dial my phone and my friend's phone
succesfully with and without the 1
On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas wrote:
Please paste the call file content (with the number 'ed of course) and
the Dial section from extensions.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, March 18, 2009 6:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)
This has to be a bug, because I dont know what else to try here
On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno wrote:
Nope, I always dial 1 + 10 digits for all my numbers. It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call file it does not work on some numbers mostly. That is the weirdest
thing I have ever seen. I tried different codecs in the call file, I still
get the PROGRESS with cause code 127
On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg
wrote:
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote:
> I have a weird problem with call using my T1 card. I can make calls fine
> using my analog and IP phones, but when I try to initiate a call using a
> .call file, I get the following error
> -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> -- Requested transfer capability: 0x00 - SPEECH
> -- PROGRESS with cause code 127 received
> it happens on certain numbers I dial, but if I dial that same number with
an
> ip or analog phone that use the T1 channel, the call is going through
> normally.
> Anybody knows why?
Are you doing anything silly with prefixing or short-circuit dialing?
in other words..
You dial 8 for an outside line, then 1+10 digits
and you're forgetting to do that for some numbers?
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Re: [asterisk-users] T1 problem (call using a .call file)
>>> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote: I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. On Wed, 18 Mar 2009, Pascal Bruno wrote: > This has to be a bug, because I dont know what else to try here Your logic is flawed. I'm a 1.2 Luddite, but it may help if you ramp up logging in logger.conf like: console = debug,dtmf,error,event,info,notice,verbose,warning And then (after "logger reload") capture the log from both a "dialed" and a "call file" call. Also, what does the call file look like? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
Here is what my extensions.conf file has:
exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _NXXNXX,n,Hangup()
exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1NXXNXX,n,Hangup()
Using the phone, I can dial any numbers succesfully.
And here is my call file:
Channel: DAHDI/g1/1XX
Callerid: XX
MaxRetries: 1
RetryTime: 5
WaitTime: 60
Context: test
Extension: s
Priority: 1
with the call file I can dial my cellphone which begin with 754XXX
but when I call my friend's cellphone from new york which is 201XXX i
get progress code 127 as follows
-- Attempting call on DAHDI/g1/1201XXX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received
I tried with the prefix 1 and without the prefix 1 it is always the same
thing, but with the handset I dial my phone and my friend's phone
succesfully with and without the 1
On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas wrote:
> Please paste the call file content (with the number ’ed of course)
> and the Dial section from extensions.conf.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Wednesday, March 18, 2009 6:24 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
>
>
>
> This has to be a bug, because I dont know what else to try here
>
>
>
>
>
> On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno wrote:
>
> Nope, I always dial 1 + 10 digits for all my numbers. It works on all
> numbers when I am using my phone (Analogue or IP) but when I do it using a
> .call file it does not work on some numbers mostly. That is the weirdest
> thing I have ever seen. I tried different codecs in the call file, I still
> get the PROGRESS with cause code 127
>
>
>
>
>
>
>
>
>
> On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg
> wrote:
>
> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote:
> > I have a weird problem with call using my T1 card. I can make calls fine
> > using my analog and IP phones, but when I try to initiate a call using a
> > .call file, I get the following error
> > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- PROGRESS with cause code 127 received
> > it happens on certain numbers I dial, but if I dial that same number with
> an
> > ip or analog phone that use the T1 channel, the call is going through
> > normally.
> > Anybody knows why?
>
> Are you doing anything silly with prefixing or short-circuit dialing?
>
> in other words..
>
> You dial 8 for an outside line, then 1+10 digits
> and you're forgetting to do that for some numbers?
>
> ___
> -- 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
>
>
>
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] T1 problem (call using a .call file)
Please paste the call file content (with the number 'ed of course) and the Dial section from extensions.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Wednesday, March 18, 2009 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 problem (call using a .call file) This has to be a bug, because I dont know what else to try here On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno wrote: Nope, I always dial 1 + 10 digits for all my numbers. It works on all numbers when I am using my phone (Analogue or IP) but when I do it using a .call file it does not work on some numbers mostly. That is the weirdest thing I have ever seen. I tried different codecs in the call file, I still get the PROGRESS with cause code 127 On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg wrote: On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote: > I have a weird problem with call using my T1 card. I can make calls fine > using my analog and IP phones, but when I try to initiate a call using a > .call file, I get the following error > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) > -- Requested transfer capability: 0x00 - SPEECH > -- PROGRESS with cause code 127 received > it happens on certain numbers I dial, but if I dial that same number with an > ip or analog phone that use the T1 channel, the call is going through > normally. > Anybody knows why? Are you doing anything silly with prefixing or short-circuit dialing? in other words.. You dial 8 for an outside line, then 1+10 digits and you're forgetting to do that for some numbers? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
This has to be a bug, because I dont know what else to try here On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno wrote: > Nope, I always dial 1 + 10 digits for all my numbers. It works on all > numbers when I am using my phone (Analogue or IP) but when I do it using a > .call file it does not work on some numbers mostly. That is the weirdest > thing I have ever seen. I tried different codecs in the call file, I still > get the PROGRESS with cause code 127 > > > > > > On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg wrote: > >> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote: >> > I have a weird problem with call using my T1 card. I can make calls >> fine >> > using my analog and IP phones, but when I try to initiate a call using a >> > .call file, I get the following error >> > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) >> > -- Requested transfer capability: 0x00 - SPEECH >> > -- PROGRESS with cause code 127 received >> > it happens on certain numbers I dial, but if I dial that same number >> with an >> > ip or analog phone that use the T1 channel, the call is going through >> > normally. >> > Anybody knows why? >> >> Are you doing anything silly with prefixing or short-circuit dialing? >> >> in other words.. >> >> You dial 8 for an outside line, then 1+10 digits >> and you're forgetting to do that for some numbers? >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
Nope, I always dial 1 + 10 digits for all my numbers. It works on all numbers when I am using my phone (Analogue or IP) but when I do it using a .call file it does not work on some numbers mostly. That is the weirdest thing I have ever seen. I tried different codecs in the call file, I still get the PROGRESS with cause code 127 On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg wrote: > On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote: > > I have a weird problem with call using my T1 card. I can make calls fine > > using my analog and IP phones, but when I try to initiate a call using a > > .call file, I get the following error > > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) > > -- Requested transfer capability: 0x00 - SPEECH > > -- PROGRESS with cause code 127 received > > it happens on certain numbers I dial, but if I dial that same number with > an > > ip or analog phone that use the T1 channel, the call is going through > > normally. > > Anybody knows why? > > Are you doing anything silly with prefixing or short-circuit dialing? > > in other words.. > > You dial 8 for an outside line, then 1+10 digits > and you're forgetting to do that for some numbers? > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote: > I have a weird problem with call using my T1 card. I can make calls fine > using my analog and IP phones, but when I try to initiate a call using a > .call file, I get the following error > -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) > -- Requested transfer capability: 0x00 - SPEECH > -- PROGRESS with cause code 127 received > it happens on certain numbers I dial, but if I dial that same number with an > ip or analog phone that use the T1 channel, the call is going through > normally. > Anybody knows why? Are you doing anything silly with prefixing or short-circuit dialing? in other words.. You dial 8 for an outside line, then 1+10 digits and you're forgetting to do that for some numbers? ___ -- 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
