[asterisk-users] T1 problem (call using a .call file)
Pascal Bruno
tipascal at gmail.com
Thu Mar 19 15:23:17 CDT 2009
Here is what I get from the console with the call file:
-- Attempting call on DAHDI/g1/1201XXXXXXX for s at fortest:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received
[Mar 19 16:12:47] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
[Mar 19 16:12:52] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
[Mar 19 16:12:57] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
[Mar 19 16:13:02] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying retry
since we're currently running '/var/spool/asterisk/outgoing/dahdi03'
-- Hungup 'DAHDI/1-1'
And here is what I get from using my analog phone:
-- Starting simple switch on 'DAHDI/32-1'
[Mar 19 16:23:50] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '1'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:50] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '1' on DAHDI/32-1
[Mar 19 16:23:50] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '1' on DAHDI/32-1
[Mar 19 16:23:50] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '2'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:50] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '2' on DAHDI/32-1
[Mar 19 16:23:50] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '2' on DAHDI/32-1
[Mar 19 16:23:51] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '0'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:51] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '0' on DAHDI/32-1
[Mar 19 16:23:51] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '0' on DAHDI/32-1
[Mar 19 16:23:51] DTMF[11266]: channel.c:2553 __ast_read: DTMF end '1'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:51] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '1' on DAHDI/32-1
[Mar 19 16:23:51] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '1' on DAHDI/32-1
[Mar 19 16:23:52] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:52] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '4' on DAHDI/32-1
[Mar 19 16:23:52] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '4' on DAHDI/32-1
[Mar 19 16:23:52] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:52] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '5' on DAHDI/32-1
[Mar 19 16:23:52] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '5' on DAHDI/32-1
[Mar 19 16:23:53] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:53] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '8' on DAHDI/32-1
[Mar 19 16:23:53] DTMF[11266]: channel.c:2619 __ast_read: DTMF end
passthrough '8' on DAHDI/32-1
[Mar 19 16:23:53] DTMF[11266]: channel.c:2553 __ast_read: DTMF end 'X'
received on DAHDI/32-1, duration 0 ms
[Mar 19 16:23:53] DTMF[11266]: channel.c:2608 __ast_read: DTMF end accepted
without begin '3' on DAHDI/32-1
[Mar 19 16:23:53] 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
[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 '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 [1201XXXXXXX at boxout:1] Dial("DAHDI/32-1",
"DAHDI/g1/1201XXXXXXX") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/1201XXXXXXX
-- 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 <danny at debsinc.com> wrote:
> Try this call file – replace XXX with your number and YYY with a valid
> SIP exten on your system
>
>
>
> Channel: DAHDI/g1/1XXXXXXXXXX
> Callerid: SIP/YYY
>
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _NXXNXXXXXX,n,Hangup()
>
>
>
> exten => _1NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN})
> exten => _1NXXNXXXXXX,n,Hangup()
>
>
>
> Using the phone, I can dial any numbers succesfully.
>
>
>
> And here is my call file:
>
>
>
> Channel: DAHDI/g1/1XXXXXXXXXX
> Callerid: XXXXXXXXXX
> MaxRetries: 1
> RetryTime: 5
> WaitTime: 60
> Context: test
> Extension: s
> Priority: 1
>
>
>
> with the call file I can dial my cellphone which begin with 754XXXXXXX
>
> but when I call my friend's cellphone from new york which is 201XXXXXXX i
> get progress code 127 as follows
>
>
>
> -- Attempting call on DAHDI/g1/1201XXXXXXX for s at 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 <danny at debsinc.com> wrote:
>
> Please paste the call file content (with the number XXXX’ed of course) and
> the Dial section from extensions.conf.
>
>
> ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 <tipascal at gmail.com> 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 <dbackeberg at gmail.com>
> wrote:
>
> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno <tipascal at gmail.com> 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/g1/1XXXXXXXXXX for s at 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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