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.
On Thu, Mar 19, 2009 at 4:23 PM, Pascal Bruno <[email protected]> wrote: > Here is what I get from the console with the call file: > > -- Attempting call on DAHDI/g1/1201XXXXXXX for s...@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 [1201xxxx...@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 <[email protected]>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:* [email protected] [mailto: >> [email protected]] *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...@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 <[email protected]> >> wrote: >> >> Please paste the call file content (with the number XXXX’ed of course) and >> the Dial section from extensions.conf. >> >> >> ------------------------------ >> >> *From:* [email protected] [mailto: >> [email protected]] *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 <[email protected]> >> 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 <[email protected]> >> wrote: >> >> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno <[email protected]> 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...@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 >> >> >> >> _______________________________________________ >> -- 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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