[asterisk-users] (CALL FILES to Local Channel)billsec Zero in cdr via cdr_adaptive_odbc
Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it Asterisk11. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database mydatabase) via cdr_adaptive_odbc. The SIP/A221 is another asterisk machine named it Elastix24. I have two BIG QUESTIONs about cdr_adaptive_odbc. First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0. Second, if I don't want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:77? I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:77(my callback exten). I can't find two cdr records in mycdr table. mysql select * from gvl_cdr; +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:37:01 | | |77 | from-internal-out-7 | Local/77@from-internal-out-7-;2 | SIP/A221- | Dial| SIP/A221/77,30 | 17 | 0 | ANSWERED| 3 | | | 1389163021.1 | 1389163021.0 | 1| | 77 | | 7 | Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the 3th one). mysql select * from gvl_cdr; +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel| dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:34:04 || | 77| from-internal-out-7 | Local/77@from-internal-out-7-;2| SIP/A221- | Dial| SIP/A221/77,30| 15 | 0 | ANSWERED|3 | | | 1389162844.1 | 1389162844.0 | 1| | 77 | | 7 | | 2014-01-08 14:34:04 | device 1000| 1000| 77| from-6 | Local/77@from-internal-out-7-;1| | ForkCDR | | 20 | 5 | ANSWERED|3 | | | 1389162844.0 | 1389162844.0 | 0| | 77 | | 7 | | 2014-01-08 14:34:24 | device 77 | 77 | 77| from-6 | Local/77@from-internal-out-7-;1| | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 | 0 | NO ANSWER |3 | | | 1389162844.0 | 1389162844.0 | 3| | | | 0 | - /var/spool/asterisk/outgoing/77.call Channel:Local/77@from-internal-out-7 WaitTime:30 Context:from-6 Extension:77 Priority:1 Set:CLID= Set:EXT=77 Set:USERID=7 --
Re: [asterisk-users] Call files without permission for asterisk to read
You could save the call file initially to /var/spool/asterisk/tmp, then adjust the permissions as needed and necessary. Finally copy the call file into the outgoing directory. This also minimizes the chance that Asterisk tries to execute a partial file, although I don't know whether one still has to take care of issues like that. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files Especially the parts about creating the files in a different directory and the parts about the scheduling call files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham Sent: Thursday, November 21, 2013 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call files without permission for asterisk to read Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
On Thu, 21 Nov 2013, Rizwan Hisham wrote: Hi all,I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? When you activate the secondary, 'touch' the files in the spool directory. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
Have you tried to restart asterisk after setting the correct permissions? HTH, Ioan On Thu, Nov 21, 2013 at 6:04 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
Looking at Eric Wieling's response and the wiki entry he mentioned, the precaution is still necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files without permission for asterisk to read
Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? I need to continue the execution of call files on secondary server if primary server fails. The call files are suppose to retry for 45 mins if the call does not get connected. Thanks in advance. -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
On Thu, 21 Nov 2013, jg wrote: Finally copy the call file into the outgoing directory. This also minimizes the chance that Asterisk tries to execute a partial file... 'mv' not 'cp' Also, create the file on the same filesystem as the spool directory so 'mv' isn't silently 'promoted' to 'cp.' -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files without permission for asterisk to read
Thanks for the responses. Touching a file after setting permissions does not work. Asterisk only looks at the new file only, not all the files in the directory. Restarting asterisk does work, but dont want to do this. Best way i think would be, as suggested by JG, to sync in a tmp directory and at the time of switch-over mv to outgoing directory. Cheers On Fri, Nov 22, 2013 at 12:36 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 21 Nov 2013, Rizwan Hisham wrote: Hi all,I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this problem? When you activate the secondary, 'touch' the files in the spool directory. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files and spool directiory shared amongst several asterisk servers
As far as I know the linux kernel uses inotify to give Asterisk a hint, that a new call file is available. Does inotify work in your environment (external storage device) at all? Am 18.11.2011 11:29, schrieb Ishfaq Malik: We have a number of asterisk servers that share a spool directory on an external storage device (for call recording). We don't use call files but now are about to just purely for our own reporting purposes. Has anyone got any experience on the behaviour of using call files when several asterisk servers share a single spool directory? We are using 1.8 Thanks Ish -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files and spool directiory shared amongst several asterisk servers
We have a number of asterisk servers that share a spool directory on an external storage device (for call recording). We don't use call files but now are about to just purely for our own reporting purposes. Has anyone got any experience on the behaviour of using call files when several asterisk servers share a single spool directory? We are using 1.8 Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 04:32:09PM +0100, Roger Burton West wrote: On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. Or better: run it under strace and verify it does. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .call files in /var/spool/asterisk/outgoing
Hi !
I have a python script that create and move .call files to
/var/spool/asterisk/outgoing
Sometimes...(in this case after 500 successfull calls) Asterisk don´t make
the calls and the .call files are in the outgoing forever...
Any Ideas?
I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior)
In my python script I move .call files using ...
import shutil
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Yes, same server, same filesystem...
On Fri, Aug 12, 2011 at 12:26 PM, Roger Burton West ro...@firedrake.orgwrote:
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Another thought - when a call in /V/S/A/O fails, the file gets appended with call info and retry occurs. You might want to write a second Python script to check for and possibly purge failed call files. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Friday, August 12, 2011 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Hello, Check if file is owned by asterisk user. Also, don't directly create in to /var/spool/asterisk/outgoing/ Create in somewhere else first and then move file to outgoing folder. Good luck. On Fri, Aug 12, 2011 at 7:09 PM, Danny Nicholas da...@debsinc.com wrote: Another thought – when a call in /V/S/A/O fails, the file gets appended with call info and retry occurs. You might want to write a second Python script to check for and possibly purge failed call files. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Friday, August 12, 2011 11:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing ** ** I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing* *** On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Sun, May 22, 2011 at 07:05:45PM -0400, Thomas Perron wrote: This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? Only on Windows (practically). If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. I assume you know what you're doing and this is for a good cause. Use the Asterisk Manager Interface. http://www.voip-info.org/wiki/view/Asterisk+manager+API Specifically, the Originate command. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
I would rather write a new bash script for text and file handing. I think you can install MONO and run windows stuff... from .net to vbs On 23 May 2011 08:09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, May 22, 2011 at 07:05:45PM -0400, Thomas Perron wrote: This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? Only on Windows (practically). If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. I assume you know what you're doing and this is for a good cause. Use the Asterisk Manager Interface. http://www.voip-info.org/wiki/view/Asterisk+manager+API Specifically, the Originate command. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Monday 23 May 2011, Thomas Perron wrote: This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? AFAIK there is no Linux interpreter for VBS :( But the format of the callfile is independent of the language used to create it -- anything else would violate the Principle of Equivalence. Just learn Perl or Python instead. Both these interpreters are installed by default on every modern Linux system. Python is for the young and trendy, and I can't get to grips with it myself. Perl is the rusty old Ford Transit van of programming languages: it may not be much to look at, but it gets the job done. And its regular expression handling is second to none. If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. You have to write a program, in whatever language you like (bash even, if you're feeling sufficiently masochistic), which generates a callfile to establish the call you want to set up. (It's also best if you generate the file in some temporary location, then move it to the intended destination directory.) -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files .vbs
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. Thank you Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
Hi Doug, Yes. I have sorted that part out. Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! Thanks On Sun, May 22, 2011 at 8:37 PM, Doug Lytle supp...@drdos.info wrote: Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Sun, 22 May 2011, Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Virtual machines or Wine may have some possibilities. I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. Write a script that executes on the Asterisk box. Where do the 10,000 numbers come from? Executing a script on the Asterisk box will enable you to monitor the status of the process better. Like only dumping xxx scripts at a time into the spool directory and sending you an email if Asterisk stops processing them for some reason. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Sun, 22 May 2011, Thomas Perron wrote: Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! pscp is a program, not a function. Part of or related to putty as I remember. Not a good idea. One of the 'bugaboos' of call files is that you are supposed to create the files in a temporary directory and move them into the spool directory. Also, you will have limited error detection ability if you are only dumping files 'willy-nilly.' Much better to do it all on the Asterisk host. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
I'm the original author of said VB Script. Steve is right, I had lots of errors - related to the fact that asterisk watches it too closely and reads the files even before they are complete - and have since updated it that it first dumps it to a temp directory, then use a bash script on the linux machine that moves all files from the temp directory to the call directory using plink. Both pscp and plink are windoz programs that utilize ssh for their functions. Pscp xfers files, and plink executes any remote commands. In the newer version pscp in the VB Script dumps it to /root/calltemps/ and /root/mvcallfiles.sh moves the files from /root/calltemps/* to /var/spool/asterisk/outgoing/ change this line: strcmd=C:\pscp -pw password c:\direcotry\strcnt\* root@asterisk:/var/spool/asterisk/outgoing to: strcmd=C:\pscp -pw password c:\directory\strcnt\* root@asterisk:/root/calltemps make sure the dir exists then add: Set objShell2 = CreateObject(WScript.Shell) strcmd2=C:\plink -pw password root@asterisk /root/mvcallfiles.sh objShell2.Run strcmd2 /root/movcallfiles.sh: #/bin/bash mv /root/calltemps/* /var/spool/asterisk/outgoing/ Hope this helps. On Sun, May 22, 2011 at 8:55 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 22 May 2011, Thomas Perron wrote: Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! pscp is a program, not a function. Part of or related to putty as I remember. Not a good idea. One of the 'bugaboos' of call files is that you are supposed to create the files in a temporary directory and move them into the spool directory. Also, you will have limited error detection ability if you are only dumping files 'willy-nilly.' Much better to do it all on the Asterisk host. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
Hello,
Thanks for replying.
Answers below:
On 23 April 2011 18:29, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.comwrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context: agents
Extension: 888210332450
Set: __PARTNER=ZonNew-Outbound
Set: NUMBER=210332450
-
In Local/210332450@ZonNew-Outbound I Set(bla='blabla');
It seems I cannot re-use this var in extension _888X in context
agents...
Basically the Channel dialplan has a Queue() and in _888X I would
like to know the peer (or interface) that answered it... What can I do?
Thanks in advance
I'm a little confused by It Seems I cannot re-use this var in extension
_888XX in context agentsOf course you can use it...but if you
set bla to a different value in your code where your callfile is processed,
Asterisk will (rightfully so) just set bla = to whatever you set it to
Now, if the callfile doesn't send a channel through the context that
you're trying to set blah, that's a little odd...
Now, as far as retrieving the information about the interface that answered
the calllook in queues.conf.samplethere's a nifty configuration
option:
*setinterfacevar=no ; (the default is no)*
Yes, I am aware of this and I do use it. However, I cannot use
MEMBERINTERFACE variable in dialplan _888X, and that is where I'm
needing it.
Also seems that its two channel legs and the only way would be to use
IMPORT() o SHARED() and for that I would have to know the channel name...
I am right now using IMPORT() like:
Set(CALLERID(num)=${IMPORT(${CHANNEL:0:$[${LEN(${CHANNEL})} -
1]}2,MEMBERNAME)});
but I fee that it is a ugly fix. What if call leg changes from 2 to 3?
That option, when set to yes, causes several variables to be created *just
* prior to the caller being bridged with the queue member...
--
Sherwood McGowan
Telecommunications and VOIP Consultant
--
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[asterisk-users] call files
Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
Hi, Using DumpChan(); Seems that Channel (where the call goes first) is a sub-channel of Context/Extension (where the call goes on CONNECT) ?? first I have: Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2: Then after: Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;1: Help ? On 23 April 2011 17:20, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance I'm a little confused by It Seems I cannot re-use this var in extension _888XX in context agentsOf course you can use it...but if you set bla to a different value in your code where your callfile is processed, Asterisk will (rightfully so) just set bla = to whatever you set it to Now, if the callfile doesn't send a channel through the context that you're trying to set blah, that's a little odd... Now, as far as retrieving the information about the interface that answered the calllook in queues.conf.samplethere's a nifty configuration option: *setinterfacevar=no ; (the default is no)* That option, when set to yes, causes several variables to be created *just*prior to the caller being bridged with the queue member... -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files or AMI originate for mass outbound call
Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files or AMI originate for mass outbound call
On 11-04-20 12:20 PM, Adolphe Cher-Aime wrote: In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? I prefer using the AMI as I have better call control. I also get to monitor the AMI events are react to them. Recently I've been using Python and starpy (twisted). -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files or AMI originate for mass outbound call
Thank you for your answer. I also prefer AMI for its flexibility. However, i have an application developped in PHP used to make more than 10 calls a day by group of 120 concurrent calls. My problem with AMI is that client keeps disconnected to AMI server. I use astmanproxy as proxy server. Do you to use have such problem with your applications ? Regards On Wed, Apr 20, 2011 at 3:11 PM, Paul Belanger pabelan...@digium.comwrote: On 11-04-20 12:20 PM, Adolphe Cher-Aime wrote: In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? I prefer using the AMI as I have better call control. I also get to monitor the AMI events are react to them. Recently I've been using Python and starpy (twisted). -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files or AMI originate for mass outbound call
On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote: Thank you for your answer. I also prefer AMI for its flexibility. However, i have an application developped in PHP used to make more than 10 calls a day by group of 120 concurrent calls. My problem with AMI is that client keeps disconnected to AMI server. I use astmanproxy as proxy server. Do you to use have such problem with your applications ? Not really, twisted (specifically the ClientFactory) has functions to handle disconnection / reconnection. It is transparent to my application, so if the client is disconnected from Asterisk, and event is fired, I stop processing calls, then wait for the client to reconnect to the AMI. Once reconnected, I begin again. [1] http://twistedmatrix.com/documents/current/core/howto/clients.html#auto4 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files or AMI originate for mass outbound call
Thanks Paul, I will take a look at twisted i will let you know. Regards On Wed, Apr 20, 2011 at 5:38 PM, Paul Belanger pabelan...@digium.comwrote: On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote: Thank you for your answer. I also prefer AMI for its flexibility. However, i have an application developped in PHP used to make more than 10 calls a day by group of 120 concurrent calls. My problem with AMI is that client keeps disconnected to AMI server. I use astmanproxy as proxy server. Do you to use have such problem with your applications ? Not really, twisted (specifically the ClientFactory) has functions to handle disconnection / reconnection. It is transparent to my application, so if the client is disconnected from Asterisk, and event is fired, I stop processing calls, then wait for the client to reconnect to the AMI. Once reconnected, I begin again. [1] http://twistedmatrix.com/documents/current/core/howto/clients.html#auto4 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Files, Variable passing
Try Set instead of SetVar. On Sat, Feb 12, 2011 at 9:59 PM, Dan Dan dani.mani...@gmail.com wrote: Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: 3001\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n); fputs($oSocket, MaxRetries: $strMaxReTry\r\n); fputs($oSocket, RetryTime: $strRetryTime\r\n); fputs($oSocket, SetVar: DIAL1=$number1\r\n); fputs($oSocket, SetVar: DIAL2=$number2\r\n); fputs($oSocket, SetVar: AcceptParallel=$ap\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); Here I am trying to set three variables but they do not seem to be passed on to the extensions for dialing Am I following the right syntax ? Thanks -dani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Files, Variable passing
Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: 3001\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n); fputs($oSocket, MaxRetries: $strMaxReTry\r\n); fputs($oSocket, RetryTime: $strRetryTime\r\n); fputs($oSocket, SetVar: DIAL1=$number1\r\n); fputs($oSocket, SetVar: DIAL2=$number2\r\n); fputs($oSocket, SetVar: AcceptParallel=$ap\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); Here I am trying to set three variables but they do not seem to be passed on to the extensions for dialing Am I following the right syntax ? Thanks -dani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files error
Adding /n partly solved the problem. The two calls are getting connected,
but LCR is not working. The 2nd call goes out on the same trunk as the first
call ( 1st call was landline, 2nd was mobile, two different routes )
Tamas
2011/2/8 fai...@vopium.com
Just verified I faced the same issue once and got it reolved by adding /n
like
Channel: Local/0036701234567@CustomCallOut-1/n in you case.
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 8:49am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters in
dial-plan as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten = _0X.,1,Set(CDR(azonosito)=${azon1})
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid1})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
[CustomCallOut-2]
exten = _0X.,1,Wait(1)
; set custom CDR
exten = _0X.,2,Set(CDR(azonosito)=${azon2})
exten = _0X.,3,Playtones(ring)
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid2})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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[asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten = _0X.,1,Set(CDR(azonosito)=${azon1})
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid1})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
[CustomCallOut-2]
exten = _0X.,1,Wait(1)
; set custom CDR
exten = _0X.,2,Set(CDR(azonosito)=${azon2})
exten = _0X.,3,Playtones(ring)
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid2})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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Re: [asterisk-users] Call files error
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten = _0X.,1,Set(CDR(azonosito)=${azon1})
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid1})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
[CustomCallOut-2]
exten = _0X.,1,Wait(1)
; set custom CDR
exten = _0X.,2,Set(CDR(azonosito)=${azon2})
exten = _0X.,3,Playtones(ring)
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid2})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
However the two calls are placed, the CDRs and the callerids are set correctly,
we can't hear each other. As I saw in the logs, the problem is that the calls
are placed in the same context, and not being connected ( like one call, but
with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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Re: [asterisk-users] Call files error
However the two calls are placed, the CDRs and the callerids are set correctly, we can't hear each other. As I saw in the logs, the problem is that the calls are placed in the same context, and not being connected ( like one call, but with the variable EXTEN changed ). I'm really confused about doing this, so can you please advise? Thanks, Tamas Tamas, Try appending /n to both of your Local channel definitions... literally a forward slash and a lowercase n...not newline :D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files error
This is obvious for the first Channel ( Channel: Local/0036701234567@CustomCallOut-1/n ), but how to set on the other party? I tried with Context: CustomCallOut-2/n but didn't worked. 2011/2/8 Sherwood McGowan sherwood.mcgo...@gmail.com However the two calls are placed, the CDRs and the callerids are set correctly, we can't hear each other. As I saw in the logs, the problem is that the calls are placed in the same context, and not being connected ( like one call, but with the variable EXTEN changed ). I'm really confused about doing this, so can you please advise? Thanks, Tamas Tamas, Try appending /n to both of your Local channel definitions... literally a forward slash and a lowercase n...not newline :D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters in
dial-plan as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten = _0X.,1,Set(CDR(azonosito)=${azon1})
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid1})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
[CustomCallOut-2]
exten = _0X.,1,Wait(1)
; set custom CDR
exten = _0X.,2,Set(CDR(azonosito)=${azon2})
exten = _0X.,3,Playtones(ring)
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid2})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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Re: [asterisk-users] Call files error
Just verified I faced the same issue once and got it reolved by adding /n like
Channel: [mailto:Local/0036701234567@CustomCallOut-1/n]
Local/0036701234567@CustomCallOut-1/n in you case.
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 8:49am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 [mailto:fai...@vopium.com] fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: Tamás Dajka [mailto:tda...@gmail.com] tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: [mailto:asterisk-users@lists.digium.com] asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten = _0X.,1,Set(CDR(azonosito)=${azon1})
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid1})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
[CustomCallOut-2]
exten = _0X.,1,Wait(1)
; set custom CDR
exten = _0X.,2,Set(CDR(azonosito)=${azon2})
exten = _0X.,3,Playtones(ring)
exten = _0X.,n,Set(CALLERPRES()=allowed)
exten = _0X.,n,Set(CALLERID(number)=${callid2})
exten = _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include = from-internal
However the two calls are placed, the CDRs and the callerids are set correctly,
we can't hear each other. As I saw in the logs, the problem is that the calls
are placed in the same context, and not being connected ( like one call, but
with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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[asterisk-users] .call files with application/data are not generating correct CDR
Hi, The exact problem that I'm experiencing is described at http://www.spinics.net/lists/asterisk/msg122364.html in an earlier posting to the mailing list, but I could find no replies to it. I installed Asterisk using Ubuntu's apt-get and then fixed the mysql conf (which doesn't load if you use the default apt-get install asterisk-mysql) by building it from scratch. I'm using Asterisk as an automated voice messaging system so need to be able to dynamically make .call files which point to different mp3 files. My calls are now being logged to the mysql database but even if I answer a call it still logs as Not Answered with a duration of zero. Setting unanswered to either yes or no makes no difference in cdr.conf - the call is still logged as Not Answered if I pick it up. Really the only way around this I can see is to check the lastapp field instead of the disposition. Lastapp is set to Dial if the call was really not answered and MP3Player if the call was answered. I see that there is a known bug in Asterisk and it is suggested to use extension.conf to set up a context rather than using call files. The problem is that I need to be able to change the MP3 that is played. Has anybody managed to solve this problem? Thanks, Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files with application/data are not generating correct CDR
You often don't get cdrs or at least useful ones unless you run the call files through a Local channel You maybe already doing this Can you check the Master.csv and see if it also is recorded incorrectly there. Is this just an issue with mysql cdrs or something else. In my setups which use freepbx I haven't had an issue with cdrs and call files if using Local channels to call Cheers Duncan On 23/08/2010, at 2:11 AM, Andy Beak wrote: Hi, The exact problem that I'm experiencing is described at http://www.spinics.net/lists/asterisk/msg122364.html in an earlier posting to the mailing list, but I could find no replies to it. I installed Asterisk using Ubuntu's apt-get and then fixed the mysql conf (which doesn't load if you use the default apt-get install asterisk-mysql) by building it from scratch. I'm using Asterisk as an automated voice messaging system so need to be able to dynamically make .call files which point to different mp3 files. My calls are now being logged to the mysql database but even if I answer a call it still logs as Not Answered with a duration of zero. Setting unanswered to either yes or no makes no difference in cdr.conf - the call is still logged as Not Answered if I pick it up. Really the only way around this I can see is to check the lastapp field instead of the disposition. Lastapp is set to Dial if the call was really not answered and MP3Player if the call was answered. I see that there is a known bug in Asterisk and it is suggested to use extension.conf to set up a context rather than using call files. The problem is that I need to be able to change the MP3 that is played. Has anybody managed to solve this problem? Thanks, Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files in 1.6
I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user Any thoughts. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files in 1.6
Jerry Geis wrote: I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user I noticed the same thing - i think something about the permissions has changed, because when I set it to read=all, write=all, it started working again. Haven't dug around enough to find out exactly what's up though. -- Jon-o Addleman - http://www.redowl.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files in 1.6
I noticed the same thing - i think something about the permissions has changed, because when I set it to read=all, write=all, it started working again. Haven't dug around enough to find out exactly what's up though. Thanks that works for me again also. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files in 1.6
On Monday 05 April 2010 11:31:04 Jonathan Addleman wrote: Jerry Geis wrote: I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user I noticed the same thing - i think something about the permissions has changed, because when I set it to read=all, write=all, it started working again. Haven't dug around enough to find out exactly what's up though. The originate command requires the originate permission. This is detailed in the UPGRADE.txt file (you _did_ read that file thoroughly, didn't you?). -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files in 1.6
Yes, so this works (maybe safer than read=all and write=all): read = system,call,command,agent,user,*originate* write = system,call,command,agent,user,*originate* I wasted probably a week on this - thanks to no documentation back in the days with v1.6. -Bruce On Mon, Apr 5, 2010 at 1:50 PM, Tilghman Lesher tles...@digium.com wrote: On Monday 05 April 2010 11:31:04 Jonathan Addleman wrote: Jerry Geis wrote: I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user I noticed the same thing - i think something about the permissions has changed, because when I set it to read=all, write=all, it started working again. Haven't dug around enough to find out exactly what's up though. The originate command requires the originate permission. This is detailed in the UPGRADE.txt file (you _did_ read that file thoroughly, didn't you?). -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files : call multiple SIP-accounts
Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3SIP/test1 for 1...@from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]: chan_sip.c:2994 create_addr: No such host: test3SIP [Mar 22 14:40:26] NOTICE[29908]: channel.c:3046 __ast_request_and_dial: Unable to request channel SIP/test3SIP/test1 [Mar 22 14:40:26] NOTICE[29908]: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) So how can I simultaneously call different SIP-accounts from a call-file ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files : call multiple SIP-accounts
Not too long ago I needed to do the same thing but apparently you need to have a separate call file for every call. The dial command didn't work with an '' separating multiple destinations. I did it through a php script running via agi. On 2010-03-22 9:56 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3SIP/test1 for 1...@from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]: chan_sip.c:2994 create_addr: No such host: test3SIP [Mar 22 14:40:26] NOTICE[29908]: channel.c:3046 __ast_request_and_dial: Unable to request channel SIP/test3SIP/test1 [Mar 22 14:40:26] NOTICE[29908]: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) So how can I simultaneously call different SIP-accounts from a call-file ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files with extensions.ael : One app must be specified
Hi, Using a 1.4 system in which dialplan is written using extensions.conf, I can use a custom .call file. On another system in which dialplan is written using extensions.ael, I can't use any custom .call file : system keeps replying : apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/toto.call When I compare both dialplans using CLI dialplan show, I don't see much differences : [ Context 'local' created by 'pbx_ael' ] (in AEL-enabled) [ Context 'local' created by 'pbx_config' ] (in non AEL-enabled) Here is the call file (I also tried commenting out Priority): Channel: SIP/700 CallerID: 692 692 MaxRetries: 1 WaitTime: 60 RetryTime: 5 Context: local Extension: 700 Priority: 1 What shall I edit to have it working ? Regards ___ -- 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] Call Files
Local channel will help you send your call through the dialplan. You can make all your decision there. If it answers, then the specified application will be execute. Check this example http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/ David Klaverstyn wrote: I have successfully created call files and I can get Asterisk to make calls based on those files. The problem I have is that it seems you need to use a Channel for the first leg of the call file. This means I have to use either a ZAP, SIP or IAX2 channel. What I would prefer to do is send the first leg of the call to a context and extension so I can send the call using DUNDi rather than a predefined channel. Once the call has been established then is should go to context, extension so the second leg of the call can be completed. Is it possible to send the first leg of a call file to DUNDi and if not aviable send over IAX2 or then ZAP? The call files seem to be limited to a channel and not allow the first leg of the call to be decided by the path of a context, extension. ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] Call Files
I have successfully created call files and I can get Asterisk to make calls based on those files. The problem I have is that it seems you need to use a Channel for the first leg of the call file. This means I have to use either a ZAP, SIP or IAX2 channel. What I would prefer to do is send the first leg of the call to a context and extension so I can send the call using DUNDi rather than a predefined channel. Once the call has been established then is should go to context, extension so the second leg of the call can be completed. Is it possible to send the first leg of a call file to DUNDi and if not aviable send over IAX2 or then ZAP? The call files seem to be limited to a channel and not allow the first leg of the call to be decided by the path of a context, extension. ___ -- 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] Call Files
On 6/11/2008 3:01 p.m., David Klaverstyn wrote: Is it possible to send the first leg of a call file to DUNDi and if not aviable send over IAX2 or then ZAP? The call files seem to be limited to a channel and not allow the first leg of the call to be decided by the path of a context, extension. Use the Local channel: http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Call files
On 14 Oct 2008, at 18:05, Christian Victor wrote: Steven Howes schrieb: Have created a system that involves using call files in the outgoing spool folder. On some occasions it retries which is fine is there any way to view calls waiting retries from the CLI? Using 1.4 btw. Have googled to no avail (although it is near the end of the day so I might be being a muppet!) One solution would be to look INTO the callfiles. The content of the file changes if there is a retry involved. I don't know by heard how exactly it changes but afair there is a line stating the retry time. AHA! I did not think of that. Sure someone said a while ago it just sucked them up and removed them. Thanks for your help. ___ -- 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] Call files
Hi All, Have created a system that involves using call files in the outgoing spool folder. On some occasions it retries which is fine is there any way to view calls waiting retries from the CLI? Using 1.4 btw. Have googled to no avail (although it is near the end of the day so I might be being a muppet!) Thanks in advance. Steve ___ -- 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] Call files
Steven Howes schrieb: Have created a system that involves using call files in the outgoing spool folder. On some occasions it retries which is fine is there any way to view calls waiting retries from the CLI? Using 1.4 btw. Have googled to no avail (although it is near the end of the day so I might be being a muppet!) One solution would be to look INTO the callfiles. The content of the file changes if there is a retry involved. I don't know by heard how exactly it changes but afair there is a line stating the retry time. Christian ___ -- 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] Call Files
Hello again.. I am working on using call files to have a form of ringback - eg if an extension is busy, the caller can dial a number and when the callee is free, the call gets made. I am trying to use a call file, which kind of works okay, however, if users have voicemail, it connects to that as opposed to waiting for the extension to become free. Is there any known way around this..? My call file looks like this: Channel: Local/[EMAIL PROTECTED] MaxRetries: 100 RetryTime: 1 WaitTime: 5 Extension: 666 Archive: yes Callerid: Callback 666 Thanks! Andy Dixon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call files with a timer?
Is there a way to set a call timer on calls created with call files? I'm looking specifically at having Asterisk hang up the call after a certain period of connection. Obviously, when I try passing an |S(time) on the channel line, I get an invalid call file... so I'm wondering if there's another way to do it. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call files with a timer?
On Jul 25, 2008, at 11:18 AM, SIP wrote:
Is there a way to set a call timer on calls created with call files?
I'm
looking specifically at having Asterisk hang up the call after a
certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I
get an
invalid call file... so I'm wondering if there's another way to do it.
N.
You could always to something like...
exten = _NXXNXX,1,Set(OutboundNumber=${EXTEN})
exten = _NXXNXX,n,goto(s,1)
exten = s,1,Dial(SIP/[EMAIL PROTECTED]|60|
L(1080:6:3))
may not be pretty, but is quick dirty ;)
Fred Posner
[EMAIL PROTECTED]
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Call files with a timer?
SIP wrote: Is there a way to set a call timer on calls created with call files? I'm looking specifically at having Asterisk hang up the call after a certain period of connection. Obviously, when I try passing an |S(time) on the channel line, I get an invalid call file... so I'm wondering if there's another way to do it. N. You should be able to put S(time) as part of the Data: line of the callfile. Something like the following: Channel: SIP/1 Application: Dial Data SIP/2||S(time) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call files with a timer?
Mark Michelson wrote: SIP wrote: Is there a way to set a call timer on calls created with call files? I'm looking specifically at having Asterisk hang up the call after a certain period of connection. Obviously, when I try passing an |S(time) on the channel line, I get an invalid call file... so I'm wondering if there's another way to do it. N. You should be able to put S(time) as part of the Data: line of the callfile. Something like the following: Channel: SIP/1 Application: Dial Data SIP/2||S(time) Oops, I left out a ':' Data: SIP/2||S(time) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call files with a timer?
That worked beautifully. Thanks, Mark. N. Mark Michelson wrote: Mark Michelson wrote: SIP wrote: Is there a way to set a call timer on calls created with call files? I'm looking specifically at having Asterisk hang up the call after a certain period of connection. Obviously, when I try passing an |S(time) on the channel line, I get an invalid call file... so I'm wondering if there's another way to do it. N. You should be able to put S(time) as part of the Data: line of the callfile. Something like the following: Channel: SIP/1 Application: Dial Data SIP/2||S(time) Oops, I left out a ':' Data: SIP/2||S(time) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] call files
I am trying to use call files that dial and play a wave file on 3 asterisk boxes console dsp. This is working. The 3 boxes are noticeably out of sync. From using 3 different call files (time to process) I'm sure is the time delay. Is there a way to get these audios more in sync? Jerry ___ -- 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] call files
Sync the clocks on your asterisk boxen using NTP or whatever, and then 'touch' the call files into the future so each asterisk waits before processing it...? Might get them closer. Another option is get all three boxes into the same meetme room, waiting a few seconds for them to be ready if you want, and play the sound file to the meetme room. Moj Jerry Geis wrote: I am trying to use call files that dial and play a wave file on 3 asterisk boxes console dsp. This is working. The 3 boxes are noticeably out of sync. From using 3 different call files (time to process) I'm sure is the time delay. Is there a way to get these audios more in sync? Jerry ___ -- 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] call files
I found the decision in using
Channel: Local/[EMAIL PROTECTED]/n
Denis V. Gudtsov пишет:
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Channel: SIP/12345
Callerid: auto 12345
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notify
Extension: 1
Priority: 1
extensions.ael follows:
context notify {
1 = {
start:
Answer();
Wait(1);
Playback(ulii_01);
HangUp();
};
I want to dial number 12345 with taking into account the dial plan,
written in context.
when i'm trying to set:
Channel: SIP/[EMAIL PROTECTED]
asterisk say's:
chan_sip.c:2737 create_addr: No such host: context
attempt to set:
Channel: SIP/context/12345
has the same result
asterisk version is 1.4.2
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[asterisk-users] call files
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Channel: SIP/12345
Callerid: auto 12345
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notify
Extension: 1
Priority: 1
extensions.ael follows:
context notify {
1 = {
start:
Answer();
Wait(1);
Playback(ulii_01);
HangUp();
};
I want to dial number 12345 with taking into account the dial plan,
written in context.
when i'm trying to set:
Channel: SIP/[EMAIL PROTECTED]
asterisk say's:
chan_sip.c:2737 create_addr: No such host: context
attempt to set:
Channel: SIP/context/12345
has the same result
asterisk version is 1.4.2
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[asterisk-users] .call files - no hangup
hi all, i have the following .call file: Channel: IAX2/[EMAIL PROTECTED]/myPOTSline MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: default Extension: 156 Priority: 1 when i drop the .call file into the /var/spool/asterisk/outgoing/ it calls out on voipjet, connects to extension 156 (which runs the a2billing AGI) and everything is great - except that if i hang up the PSTN side, nothing happens. Only when the AGI decides to hang up does it hang up. Just for reference, extension 156 in default is: exten = 156,1,Answer exten = 156,2,Wait,1 exten = 156,3,DeadAGI(a2billing.php) exten = 156,4,Hangup anyone have any idea why a hang up on the PSTN side is not being accepted? thanks, yair ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .call files no longer generating CDR files
I've got a curious one: all of a sudden my .call files and my manager API 'Originate' actions are no longer producing a CSV file. The call still generates just fine, and Master.csv is updated. However, I don't get the usual CSV file in the form of xx.csv where xx=account number. I didn't make any changes that I'm aware of. Is there something to check? I'm on 1.2.12, and this machine was working fine just a few days ago... Any insights would be much appreciated. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .Call files do not seem to work
Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/910555 CallerID : Leebo 55 MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority: 1 2. And then mv'd to /var/spool/asterisk/outgoing As I mentioned, Asterisk appears to be grabbing the file, but there is no call made. Q. Do calls originated like this show up in CLI output? Q. The context portion of the package refers to the context to place the call in after the remote person answers, right? Or is it the context that the origination should dial out on? I've tried both ways just in case, but no go. Thanks for any help. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .Call files do not seem to work
On Tue, Dec 19, 2006 at 10:17:51AM -0500, Lee wrote: Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/910555 CallerID : Leebo 55 MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority: 1 2. And then mv'd to /var/spool/asterisk/outgoing As I mentioned, Asterisk appears to be grabbing the file, but there is no call made. Q. Do calls originated like this show up in CLI output? In the CLI: sip show peer axVoice show dialplan main_menu set verbose 3 Then drop the call file What is the CLI trace of the above? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call file into /var/spool/asterisk/outgoing
did not produce any CLI output. The file was copied correctly, I
believe and not present in the /outgoing directory when I checked with a
simple ls command.
# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing
=== sip show peer axVoice ===
=
CLI
* Name : axVoice
Secret : Set
MD5Secret: Not set
Context : incoming
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
FromUser : datatrak
FromDomain : 216.143.130.36
Callgroup:
Pickupgroup :
Mailbox :
VM Extension : 555
LastMsgsSent : -1
Call limit : 0
Dynamic : No
Callerid :
Expire : -1
Insecure : port,invite
Nat : Always
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: No
DTMFmode : rfc2833
LastMsg : 0
ToHost : 216.143.130.36
Addr-IP : 216.143.130.36 Port 5060
Defaddr-IP : 216.143.130.36 Port 0
Def. Username: set
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent:
Reg. Contact :
=== show dialplan main_after_hours ===
(I mistyped the name of the context in original post)
CLI show dialplan main_after_hours
[ Context 'main_after_hours' created by 'pbx_config' ]
'1' =1. Playback(transfer)
[pbx_config]
2. Macro(DialExtenVM|111|30|tm)
[pbx_config]
3. Set(EXTEN=955)
[pbx_config]
4. GoTo(Management|955|1)
[pbx_config]
5. Playback(transfer)
[pbx_config]
6. Macro(DialExtenVM|111|30|tr)
[pbx_config]
7. Set(EXTEN=955)
[pbx_config]
8. GoTo(Management|955|1)
[pbx_config]
9. Playback(custom/no_tech_available)
[pbx_config]
10. Voicemail(111)
[pbx_config]
'2' =1.
Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
2. Goto(support_non_emergency|s|1)
[pbx_config]
'444' = 1. Set(LIMIT_PLAYAUDIO_CALLEE=yes)
[pbx_config]
2. Dial(SIP/111|30|mgL(1:1:5000))
[pbx_config]
3. Wait(3)
[pbx_config]
4. Goto(main_after_hours|s|1)
[pbx_config]
'9' =1.
Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
2. Goto(main_branch|s|1)
[pbx_config]
'i' =1. GotoIf($[ ${FAIL_MENU} != ]|?2:3)
[pbx_config]
2. Goto(${FAIL_MENU}|s|1)
[pbx_config]
3. Goto(main_branch|s|1)
[pbx_config]
's' =1. Answer()
[pbx_config]
2. Wait(1)
[pbx_config]
3. Background(custom/after_hours)
[pbx_config]
't' =1. GotoIf($[ ${TIMEOUT_MENU} != ]|?2:3)
[pbx_config]
2. Goto(${TIMEOUT_MENU}|s|1)
[pbx_config]
3. Goto(main_branch|s|1)
[pbx_config]
'_ZZZ' = 1.
Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m)
[pbx_config]
--
Warm Regards,
Lee
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call file into /var/spool/asterisk/outgoing
did not produce any CLI output. The file was copied correctly, I
believe and not present in the /outgoing directory when I checked with a
simple ls command.
# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing
Are both the current directory and /var/spool/asterisk/outgoing on the
same filesystem? If not, a 'mv' is implemented through a copy.
Anyway, you left out the CLI output of dropping trhe file.
Can Asterisk read that file? Write to it?
=== sip show peer axVoice ===
=
CLI
* Name : axVoice
Secret : Set
MD5Secret: Not set
Context : incoming
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
FromUser : datatrak
FromDomain : 216.143.130.36
Callgroup:
Pickupgroup :
Mailbox :
VM Extension : 555
LastMsgsSent : -1
Call limit : 0
Dynamic : No
Callerid :
Expire : -1
Insecure : port,invite
Nat : Always
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: No
DTMFmode : rfc2833
LastMsg : 0
ToHost : 216.143.130.36
Addr-IP : 216.143.130.36 Port 5060
Defaddr-IP : 216.143.130.36 Port 0
Def. Username: set
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent:
Reg. Contact :
=== show dialplan main_after_hours ===
(I mistyped the name of the context in original post)
CLI show dialplan main_after_hours
[ Context 'main_after_hours' created by 'pbx_config' ]
'1' =1. Playback(transfer)
[pbx_config]
2. Macro(DialExtenVM|111|30|tm)
[pbx_config]
3. Set(EXTEN=955)
[pbx_config]
4. GoTo(Management|955|1)
[pbx_config]
5. Playback(transfer)
[pbx_config]
6. Macro(DialExtenVM|111|30|tr)
[pbx_config]
7. Set(EXTEN=955)
[pbx_config]
8. GoTo(Management|955|1)
[pbx_config]
9. Playback(custom/no_tech_available)
[pbx_config]
10. Voicemail(111)
[pbx_config]
'2' =1.
Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
2. Goto(support_non_emergency|s|1)
[pbx_config]
'444' = 1. Set(LIMIT_PLAYAUDIO_CALLEE=yes)
[pbx_config]
2. Dial(SIP/111|30|mgL(1:1:5000))
[pbx_config]
3. Wait(3)
[pbx_config]
4. Goto(main_after_hours|s|1)
[pbx_config]
'9' =1.
Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
2. Goto(main_branch|s|1)
[pbx_config]
'i' =1. GotoIf($[ ${FAIL_MENU} != ]|?2:3)
[pbx_config]
2. Goto(${FAIL_MENU}|s|1)
[pbx_config]
3. Goto(main_branch|s|1)
[pbx_config]
's' =1. Answer()
[pbx_config]
2. Wait(1)
[pbx_config]
3. Background(custom/after_hours)
[pbx_config]
't' =1. GotoIf($[ ${TIMEOUT_MENU} != ]|?2:3)
[pbx_config]
2. Goto(${TIMEOUT_MENU}|s|1)
[pbx_config]
3. Goto(main_branch|s|1)
[pbx_config]
'_ZZZ' = 1.
Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m)
[pbx_config]
--
Warm Regards,
Lee
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--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work
Tzafrir Cohen wrote: On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote: In the CLI: sip show peer axVoice show dialplan main_menu set verbose 3 Then drop the call file What is the CLI trace of the above? Hi, thanks for responding. Please see the output below. Please note that moving a call file into /var/spool/asterisk/outgoing did not produce any CLI output. The file was copied correctly, I believe and not present in the /outgoing directory when I checked with a simple ls command. # cp lee.call test.call # mv test.call /var/spool/asterisk/outgoing Are both the current directory and /var/spool/asterisk/outgoing on the same filesystem? If not, a 'mv' is implemented through a copy. Anyway, you left out the CLI output of dropping trhe file. Can Asterisk read that file? Write to it? Hi, As I mentioned above, the action of dropping a .call into the /outgoing directory did not produce any CLI output. I did this through 2 putty sessions. The first, we setup to watch the CLI output and the second was to use the commandline to move the .call into the /outgoing directory. Asterisk must be doing *something* with the file (if just deleting it) because if I check the /outgoing directory after I move the file, there is no file there. It's deleted. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work
Lee wrote: As I mentioned above, the action of dropping a .call into the /outgoing directory did not produce any CLI output. I did this through 2 putty sessions. The first, we setup to watch the CLI output and the second was to use the commandline to move the .call into the /outgoing directory. set verbose 50 and try again. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk
If you are using Windows to generate the .call files, make sure they are in Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use Crimson Editor www.crimsoneditor.com to make the file, and click Document File Format Unix Format. I ran into this same problem, and it turns out my Asterisk install would not use Windows-formatted text files, it would just ignore them and delete them. -Original Message- From: Lee [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work Tzafrir Cohen wrote: On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote: In the CLI: sip show peer axVoice show dialplan main_menu set verbose 3 Then drop the call file What is the CLI trace of the above? Hi, thanks for responding. Please see the output below. Please note that moving a call file into /var/spool/asterisk/outgoing did not produce any CLI output. The file was copied correctly, I believe and not present in the /outgoing directory when I checked with a simple ls command. # cp lee.call test.call # mv test.call /var/spool/asterisk/outgoing Are both the current directory and /var/spool/asterisk/outgoing on the same filesystem? If not, a 'mv' is implemented through a copy. Anyway, you left out the CLI output of dropping trhe file. Can Asterisk read that file? Write to it? Hi, As I mentioned above, the action of dropping a .call into the /outgoing directory did not produce any CLI output. I did this through 2 putty sessions. The first, we setup to watch the CLI output and the second was to use the commandline to move the .call into the /outgoing directory. Asterisk must be doing *something* with the file (if just deleting it) because if I check the /outgoing directory after I move the file, there is no file there. It's deleted. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk
Colin Anderson wrote: If you are using Windows to generate the .call files, make sure they are in Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use Crimson Editor www.crimsoneditor.com to make the file, and click Document File Format Unix Format. I ran into this same problem, and it turns out my Asterisk install would not use Windows-formatted text files, it would just ignore them and delete them. Hi Colin, Thanks for responding. I've run into the problem elsewhere myself. Alas, I wrote the call file using nano on the linux box through ssh/putty. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk
The only other thing that comes to mind is that .call files are very sensitive to whitespace; you may have unintentially padded the .call file with whitespace or tabs that it does not like. The attached .call file works on my 1.0.9 server. Maybe it can give you some insight. -Original Message- From: Lee [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk Colin Anderson wrote: If you are using Windows to generate the .call files, make sure they are in Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use Crimson Editor www.crimsoneditor.com to make the file, and click Document File Format Unix Format. I ran into this same problem, and it turns out my Asterisk install would not use Windows-formatted text files, it would just ignore them and delete them. Hi Colin, Thanks for responding. I've run into the problem elsewhere myself. Alas, I wrote the call file using nano on the linux box through ssh/putty. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 614.call Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk
Colin Anderson wrote: The only other thing that comes to mind is that .call files are very sensitive to whitespace; you may have unintentially padded the .call file with whitespace or tabs that it does not like. The attached .call file works on my 1.0.9 server. Maybe it can give you some insight. Looking at your file, there are tabs between the pairname/separator and the actual value. The example on the wiki didn't see to use tabs and I guess that was it because it's working now...kinda. Good eye and thanks to everyone else of the help. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call files and cdr I need src different from CallerID(number)
Hi, if I dial normal with the dial comman I have in my cdr file the peer-name as source and the CALLERID (number and name) as I have set it in the dialplan. Now Iam using call files and Iam using in the file for example: Callerid: name 333 333 will be used for the field src AND the CALLERID(number) in the cdr file. So I dont have the choice to set CALLERID(number) different to the peer-name (src in the cdr file). How this can be fixed. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call files, fax
Hello, I have a couple of questions: 1) Before heading off for a bit of vacation, I was having a wierd problem where I was getting more than one call per callfile placed in the outgoing/ spool. I describe it here: http://forums.digium.com/viewtopic.php?t=3455 so far, so good - it's not doing it right now, but what might cause that? 2) app_txfax I need to know if a fax has gone through or not. My reading of txfax seems to indicate that it basically just fails, rather than giving me anything I can work with to try and fail gracefully (letting the user know that things didn't go well). Is that indeed correct? I don't know Asterisk that well, so I may be completely off base:-) What would be the best way to make it interact better with the dial plan so that one could detect if it fails and act accordingly? Set a variable? 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... Thanks! -- Webster srl Sede legale: Via del Seminario, 3 35122 Padova Sede operativa: Via S. Breda, 28 35010 Limena (PD) Tel. +39 049 652527 - Fax +39 049 655297 Email: [EMAIL PROTECTED] Visita www.libreriauniversitaria.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files, fax
David N. Welton [EMAIL PROTECTED] wrote: 2) app_txfax I need to know if a fax has gone through or not. My reading of txfax seems to indicate that it basically just fails, rather than giving me anything I can work with to try and fail gracefully (letting the user know that things didn't go well). Is that indeed correct? I don't know Asterisk that well, so I may be completely off base:-) What would be the best way to make it interact better with the dial plan so that one could detect if it fails and act accordingly? Set a variable? 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... You really should consider HylaFAX - www.hylafax.org. It has what you're missing - a fully featured queue manager / scheduler that takes care of retries for you, and notifies the sender of any failures encountered. It can be integrated with Asterisk via analog or digital lines, or by using a software-based modem such as IAXmodem. -Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files, fax
Darren Nickerson wrote: 3) I'm working on a small, simple email-fax system. Just out of curiosity, what else is out there for Asterisk? I found AsterFax, but it looks a little bit hairy to set up... You really should consider HylaFAX - www.hylafax.org. It has what you're missing - a fully featured queue manager / scheduler that takes care of retries for you, and notifies the sender of any failures encountered. It can be integrated with Asterisk via analog or digital lines, or by using a software-based modem such as IAXmodem. Hi, I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? -- Webster srl Sede legale: Via del Seminario, 3 35122 Padova Sede operativa: Via S. Breda, 28 35010 Limena (PD) Tel. +39 049 8842188 Email: [EMAIL PROTECTED] Visita www.libreriauniversitaria.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files, fax
On Mon, 2006-01-09 at 16:40 +0100, David N. Welton wrote: Hi, I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? http://sf.net/projects/iaxmodem iaxmodem connects to asterisk via iax2 (localhost interface prefered) and exposes a /dev entry suitable for use with hylafax, it even has a hylafax modem definition file to make it a little easier. all software no hardware -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call files, fax
I thought about using Hylafax, but after looking around a bit, I got the impression that it's not exactly trivial to integrate it with Asterisk, and that it will require a dedicated incoming line. Perhaps I'm mistaken? It isn't that bad basically download compile and install the trick is to find the version of HylaFax that will compile clean under your kernel. You need a version that was released about the same time as the vintage of your kernel. IAXmodem works. It provides a virtual modem that interacts with Asterisk via IAX. Otherwise, you need a channel bank that will terminate to some POTS lines and regular modems + free serial ports, but then, any other fax package requires that as well. The big weakness in Hylafax is the client. 90% of the time the client will be under Windows, and your choices are Cypheus, which is pretty and user friendly but slow and crash-y or WHFC which is ugly and nasty but works 100% and has slick features like offline faxing. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call files, fax
Colin Anderson [EMAIL PROTECTED] wrote: The big weakness in Hylafax is the client. 90% of the time the client will be under Windows, and your choices are Cypheus, which is pretty and user friendly but slow and crash-y or WHFC which is ugly and nasty but works 100% and has slick features like offline faxing. There's a few more choices than those two ;-) See: http://www.hylafax.org/content/Client_Software -Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context --ResponseTimeout the best answer?
Hi! Upgarde to 1.2.1 and try again - 1.2.0 (and maybe the beta) had a bug concerning .call files and the non-passing on of variables that might affect you as well. Cheers, Philipp Hmmm seems like every dialplan snippet I've seen so far relies on ResponseTimeout and looping back to s,1. Is this the only way I can get this to work kind-of the way I want? Any ideas welcome. Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] .call files on PRI not waiting for answer in de sired context
If I generate a .call file to an external callee through my PRI, Asterisk will not wait to execute the priority in the target context, and instead will continue on as soon as the channel is dialled. I want it to wait for an answer, THEN continue. It detects the answer correctly. I have callprogress=yes in Zapata.conf. I have read the wiki with respect to this issue. Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context
Colin Anderson wrote: Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? I'm not saying this has been fixed since that point, but why in the world are you running 1.2.0 beta 1 when 1.2.1 has been released? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] .call files on PRI not waiting for answer in desired context
Hey, baby steps. Truth is I've been too busy and I don't have a pressing need to upgrade. Everything's working fine (except this, of course!) -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context Colin Anderson wrote: Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? I'm not saying this has been fixed since that point, but why in the world are you running 1.2.0 beta 1 when 1.2.1 has been released? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] .call files on PRI not waiting for answer in de sired context --ResponseTimeout the best answer?
Hmmm seems like every dialplan snippet I've seen so far relies on ResponseTimeout and looping back to s,1. Is this the only way I can get this to work kind-of the way I want? Any ideas welcome. -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 12:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context If I generate a .call file to an external callee through my PRI, Asterisk will not wait to execute the priority in the target context, and instead will continue on as soon as the channel is dialled. I want it to wait for an answer, THEN continue. It detects the answer correctly. I have callprogress=yes in Zapata.conf. I have read the wiki with respect to this issue. Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .call files in outgoing dont get run
First, check your permissions. That seems to be the most common issue there. Make sure that the asterisk user has full read and right permissions to the directory and the file. Second check your logs. After you do this, something like: grep -i warning /var/log/asterisk/full | tail -n 10 may give you something useful to go on. Best Wishes, Chris Travers Metatron Technology Consulting ___ 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] .call files in outgoing dont get run
Hi all, Created file 1.call in /var/spool/asterisk/tmp/1.call with. -- Channel: Local/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: queue Extension: 1000 Priority: 2 -- When I move this to /var/spool/asterisk/outgoing/ it doesnt get runned nothing happens, tried on alot of diffrent ways, but nothing works. What could be wrong? daniel eriksson ___ 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] Call Files to Terminate a call to the dialplan not directly to a channel
You can use (at least in asterisk CVS), this: Channel: Local/[EMAIL PROTECTED] then in extensions.conf [from-internal] exten = 1234,1,Dial(whatever) exten = 1234,2,Dial(otherprov) Not testet though ;) Julian J. M. On 4/14/05, Mystery Glitch [EMAIL PROTECTED] wrote: Can I use the .call files to place a call using the dialplan instead of the channel directly? ---Channel: SIP/[EMAIL PROTECTED] Context: testing Extension: playsample Priority: 1 CallerID: Company 8882650946 WaitTime: 15 ___ 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
