Re: [asterisk-users] fax spandsp
Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] -- __ Edwin Lam edwin@officegeneral.com __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370 __ __ http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xEF11A895 __ -- _ -- 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] func odbc and mult iquery
Hello, Does asterisk func odbc support multi query? I'm executing stored procedure which returns two tables. With tsql command I can see both tables. But asterisk only shows the first. My database is MSSQL. Maybe there is workaround... Thanks -- Best Regards, Giedrius -- _ -- 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] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
Thanks, I have uploaded the patch to the website and will let you know the feedback we receive. Greetings, Joachim -- _ -- 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] fax spandsp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Edwin Lam skrev: Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] I am running Asterisk 1.6.2.1 and 1.6.2.2 with Spandsp 0.0.6-pre17 with the exact same error messages. But my asterisk does not have any trouble apart from the messages themselves. I did however run into this earlier, and I believe it was fixed for the 1.6.2.x series. At least it worked for me. Best regards, Tommy Botten Jensen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuXYlwACgkQ573V05EH/pZpZQCeO8FPGqAJ4cRDlnyZOERbgNoj 0TEAmgOiY0byfIy3SIM5GR9gDrG+LZEY =oN/L -END PGP SIGNATURE- -- _ -- 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] callerid change name
hi folks, I want to change the callerid= variable generated from php page. Let me explain : in /var/log/asterisk/cdr-csv/Master.csv we have the following line : device 360 we want to change it we don't want the extension name but the displayed name instead of 360 we want Poste 360 or something else I know it is automatically generated but where can I get the variable to change this ? -- Cordialement, / Greetings, Georghy FUSCO -- _ -- 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] Extensions.conf changed but not take effect
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
my dailplan is :
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten = _95040X,n(welcome),Background(${welcomefile},,123)
...
exten = i,1,Playback(invalid)
exten = i,2,Goto(${CALLINNUM},welcome)
first, i have changed _95040XX to _95040X , but when excute
exten = i,2,Goto(${CALLINNUM},welcome)
cli error message as follow:
-- Executing [...@extinvalid:2] Goto(SIP/1000-0005,
95040,_95040XX,welcome) in new stack
[Mar 10 17:17:27] NOTICE[5057]: pbx.c:3731 pbx_extension_helper:
Cannot find extension '_95040XX' in context '95040'
[Mar 10 17:17:27] WARNING[5057]: pbx.c:9602 pbx_parseable_goto:
Priority 'welcome' must be a number 0, or valid label
do you know what's wrong? does asterisk have some buffer or cache
files? although i change but read the old file?
thank you very much!
--
Best regards,
Sucan
--
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Re: [asterisk-users] Extensions.conf changed but not take effect
check the Goto cmd syntax
2010/3/10 Zhang Shukun bit...@gmail.com
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
my dailplan is :
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten = _95040X,n(welcome),Background(${welcomefile},,123)
...
exten = i,1,Playback(invalid)
exten = i,2,Goto(${CALLINNUM},welcome)
first, i have changed _95040XX to _95040X , but when excute
exten = i,2,Goto(${CALLINNUM},welcome)
cli error message as follow:
-- Executing [...@extinvalid:2] Goto(SIP/1000-0005,
95040,_95040XX,welcome) in new stack
[Mar 10 17:17:27] NOTICE[5057]: pbx.c:3731 pbx_extension_helper:
Cannot find extension '_95040XX' in context '95040'
[Mar 10 17:17:27] WARNING[5057]: pbx.c:9602 pbx_parseable_goto:
Priority 'welcome' must be a number 0, or valid label
do you know what's wrong? does asterisk have some buffer or cache
files? although i change but read the old file?
thank you very much!
--
Best regards,
Sucan
--
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Re: [asterisk-users] Extensions.conf changed but not take effect
On Wed, Mar 10, 2010 at 10:38 AM, Zhang Shukun bit...@gmail.com wrote: hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: Type dialplan reload in your CLI -- _ -- 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] I loose incoming call after transfer
Hello list. An incoming call goes to the queue. Then is routed to a free SIP-member1. When this SIP-member1 transfers the call to another SIP-member2, and this SIPmember-2 rejects the call, then the communication is lost. How can I make the call go back to the SIP-member1 ? Or maybe back to the queue ? To transfer we use the 'transfer'-button on the Grandstream/YeaLink IP-phone. Greetingz. 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] Which spandsp to use with 1.6.2?
On Tue, Mar 09, 2010 at 06:20:53PM -0500, sean darcy wrote: Receiving a fax pstn - pstn with 1.6.2.6-rc2: -- Executing [...@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack -- Executing [...@incoming-pstn-line:2] Wait(DAHDI/4-1, 3) in new stack -- Executing [...@incoming-pstn-line:3] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- Redirecting DAHDI/4-1 to fax extension -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax Detected) in new stack -- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1, incoming-fax,s,1) in new stack -- Goto (incoming-fax,s,1) -- Executing [...@incoming-fax:1] Set(DAHDI/4-1, FAXFILE=/var/spool/asterisk/fax/20100309_1259) in new stack -- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1, /var/spool/asterisk/fax/20100309_1259.tif) in new stack [Mar 9 13:02:13] WARNING[25317]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Mar 9 13:02:13] WARNING[25317]: app_fax.c:817 transmit: Transmission error The fax completes to a standard fax machine. I'm using spandsp-0.0.5, which, AFAICT, is the last release. AFAIU, the author recommends latest 0.0.6pre . pre17 in this case. There does not seem to be any separate 0.0.5 branch maintained anywhere. However I also see: http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre17.tgz and http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20100228.tar.gz This is exactly the same as spandsp-0.0.6pre16 . http://gitorious.org/spandsp/spandsp/commit/4bb278aac18528503966f2b874e42c6ab036b153 spandsp-0.0.6pre17 is spandsp-0.0.6pre16 with a fix of bug (typo?) http://gitorious.org/spandsp/spandsp/commit/fb0228a1898cbaa28f5b0956f70ada35d708090c See: http://gitorious.org/spandsp/spandsp/commits/master Sadly there's no official VCS tree available. -- 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] Passing a parameter to voicemail
Hi all, is it possible to send a parameter to the asterisk voicemail system? I would like to create an IVR that asks for a client code and after that transfer to a particular voicemail. Then the voicemail should send an email with the client code in the subject. The only problem is about the saving of client code. Do you have any hint? Thanks and regards, Carlo Dimaggio -- _ -- 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] callerid change name
Georghy a écrit :
hi folks,
I want to change the callerid= variable generated from php page.
Let me explain :
in /var/log/asterisk/cdr-csv/Master.csv we have the following line :
device 360
we want to change it we don't want the extension name but the displayed name
instead of 360 we want Poste 360 or something else
I know it is automatically generated but where can I get the variable to
change this ?
I just change to the following lines in extensions_custom.conf
[macro-dundi-priv]
exten = s,1,Macro(user-callerid)
exten = s,n,Goto(${ARG1},1)
include = dundi-priv-lookup
and it works great :)
--
Cordialement, / Greetings,
Georghy FUSCO
--
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Re: [asterisk-users] CallerID presented in Asterisk
On 10 Mar 2010, at 05:41, Gopalakrishnaiyer Venugopal-Q16770 wrote: So is this a bug in Asterisk 1.6? Has anyone verified/reported this issue? Read what people send you. Are you using FreePBX? If yes, then that ticket is a FreePBX bug report. If you read the words in the report it will explain it. Steve -- _ -- 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] Passing a parameter to voicemail
you can consider the internal db of asterisk. look at: http://www.voip-info.org/wiki/view/Asterisk+database 2010/3/10 Carlo Dimaggio jaasmail...@gmail.com Hi all, is it possible to send a parameter to the asterisk voicemail system? I would like to create an IVR that asks for a client code and after that transfer to a particular voicemail. Then the voicemail should send an email with the client code in the subject. The only problem is about the saving of client code. Do you have any hint? Thanks and regards, Carlo Dimaggio -- _ -- 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] dtmf payload 100
Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? Regards, Katerina -- _ -- 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] Extensions.conf changed but not take effect
This one is pretty clear - asterisk is expecting to go to a valid context
and tag. You are dialing 9504012345 and the goto is trying to go to
9504012345,welcome or 9504012345,3 when it should actually go to
${CALLINNUM}:0:5 to go to 95040,welcome for any call starting with 95040.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emanuele
Carbone
Sent: Wednesday, March 10, 2010 3:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extensions.conf changed but not take effect
check the Goto cmd syntax
2010/3/10 Zhang Shukun bit...@gmail.com
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
my dailplan is :
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten = _95040X,n(welcome),Background(${welcomefile},,123)
...
exten = i,1,Playback(invalid)
exten = i,2,Goto(${CALLINNUM},welcome)
first, i have changed _95040XX to _95040X , but when excute
exten = i,2,Goto(${CALLINNUM},welcome)
cli error message as follow:
-- Executing [...@extinvalid:2] Goto(SIP/1000-0005,
95040,_95040XX,welcome) in new stack
[Mar 10 17:17:27] NOTICE[5057]: pbx.c:3731 pbx_extension_helper:
Cannot find extension '_95040XX' in context '95040'
[Mar 10 17:17:27] WARNING[5057]: pbx.c:9602 pbx_parseable_goto:
Priority 'welcome' must be a number 0, or valid label
do you know what's wrong? does asterisk have some buffer or cache
files? although i change but read the old file?
thank you very much!
--
Best regards,
Sucan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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Re: [asterisk-users] Dialplan behaviour
I understand the “ex-girlfriend” situation, in fact I want to do that, the problem is when I don´t put the last line and call from 92 or 91 this don´t work. I put the ex-girlfriend exception because without this, calls from 91 and 92 don´t match their extensions. On Mon, Mar 8, 2010 at 4:52 PM, Danny Nicholas da...@debsinc.com wrote: You made the troncal-sip context into a “ex-girlfriend” situation where only calls from extensions 91 and 92 would process. When you added the last line, that made it an open context with two egf exceptions. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, March 08, 2010 1:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dialplan behaviour I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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] Dialplan behaviour
This may just be my opinion, but EG logic works best in an established call, like this [TRONCAL-SIP] exten = 225,1,answer exten=225/91,2,Answer exten=225/91,3,Echo exten=225/92,2,Answer exten=225/92,3,Playback(conf-invalid) exten=225,hangup This way, 225 is answered and hungup regardless of caller, and 91/92 get their specific handling. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Wednesday, March 10, 2010 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialplan behaviour I understand the ex-girlfriend situation, in fact I want to do that, the problem is when I don´t put the last line and call from 92 or 91 this don´t work. I put the ex-girlfriend exception because without this, calls from 91 and 92 don´t match their extensions. On Mon, Mar 8, 2010 at 4:52 PM, Danny Nicholas da...@debsinc.com wrote: You made the troncal-sip context into a ex-girlfriend situation where only calls from extensions 91 and 92 would process. When you added the last line, that made it an open context with two egf exceptions. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Monday, March 08, 2010 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dialplan behaviour I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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] dialplan reload: not working with large dialplans
Andreas Brodmann wrote: Lief, I'd be glad to receive your feedback. I don't think it's a limit of lines by itself. I haven't found any useful debug information so far, but I think the dialplan parser stumbles over something. The problem is reproduceable on different hareware, it only sticks to newer 1.6 releases. Up until 1.6.0 I haven't seen this problem with the same dialplan. When I go down to 1.6.0 I don't have this problem anymore. I'd suggest you post a bug to the issue tracker: https://issues.asterisk.org with the dialplan that reproduces this issue. I have a larger dialplan that doesn't seem to do this on my system, so there must be something unique about your dialplan that the parser is choking on. Being able to reproduce the issue will likely be critical in resolving it. Leif. -- _ -- 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] Passing a parameter to voicemail
Il giorno 10/mar/10, alle ore 14:48, Emanuele Carbone ha scritto: you can consider the internal db of asterisk. look at: http://www.voip-info.org/wiki/view/Asterisk+database What about the matching between the parameter stored in db and the voicemail message? Asterisk can be configured to use a db value in the sending mail procedure? Thanks -- _ -- 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] 00h323 cant get gatekeeper to connect
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
23:02:59:045 GkClient Received RAS Message
23:02:59:045 Received RAS Message = {
23:02:59:045 gatekeeperReject = {
23:02:59:045requestSeqNum = {
23:02:59:045 1
23:02:59:046}
23:02:59:046protocolIdentifier = {
23:02:59:046 {
23:02:59:046 0 0 8 2250 0 5 }
23:02:59:046}
23:02:59:047rejectReason = {
23:02:59:047 resourceUnavailable = {
23:02:59:048 NULL
23:02:59:048 }
23:02:59:048}
23:02:59:048 }
23:02:59:048 }
23:02:59:048 Gatekeeper Reject (GRJ) message received
23:02:59:048 Deleted GRQ Timer.
23:02:59:048 Error: Gatekeeper Reject - Resource Unavailable
23:02:59:049 Error: Gatekeeper error. Either Gk not responding or Gk
sending invalid messages
23:02:59:049 Error: Gatekeeper error detected. Closing GkClient as Gk mode
is UseSpecifcGatekeeper
23:02:59:049 Destroying Gatekeep
--
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[asterisk-users] multiple RTP port ranges for SIP
We are coordinating a connection to a SIP provider who told us they use two port ranges for RTP, 7000-8000 and 1-2. We've never encountered that before (and I believe rtp.conf only supports a single range). We can obviously setup 7000-2 within RTP.conf, but I'm wondering if there is any other significance to this config? The other end is using an asterisk server too, so how can they be using a split range like that? I believe the originator requests the ports for RTP... Thanks, MD -- _ -- 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] multiple RTP port ranges for SIP
Klaus Darilion wrote: That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow configuration of different ranges for UDPTL and RTP (although it shouldn't be a problem to configure the same ports in rtp.conf and udptl.conf) It absolutely would be a problem to have identical, or even overlapping, port ranges specified in rtp.conf and udptl.conf. Those port numbers are UDP port numbers, and they must be unique across the system for things to work properly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] multiple RTP port ranges for SIP
On 10.03.2010 16:35, Michelle Dupuis wrote: We are coordinating a connection to a SIP provider who told us they use two port ranges for RTP, 7000-8000 and 1-2. They use these ports. So there is nothing you have to do on Asterisk side to handle this, as Asterisk's RTP ports are different from your provider's RTP ports. Only if you have a restictive firewall you should open the ports used by your provider. We've never encountered that before (and I believe rtp.conf only supports a single range). We can obviously setup 7000-2 within RTP.conf, but I'm wondering if there is any other significance to this config? As said above: no. The other end is using an asterisk server too, so how can they be using a split range like that? I believe the originator requests the ports for RTP... That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow configuration of different ranges for UDPTL and RTP (although it shouldn't be a problem to configure the same ports in rtp.conf and udptl.conf) regards klaus -- _ -- 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] 00h323 cant get gatekeeper to connect
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without
any problems. I need your ooh323.conf and all relevant CM config
(signal-group, trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote: I'm trying to
connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN). When
chan_ooh323 first loads it tries to establish a connection with the gk but
I it fails. I have the following extract from the ooh323 log. Can anyone
give some insight? Thanks! MD 23:02:59:045 Sent GRQ message
23:02:59:045 GkClient Received RAS Message
23:02:59:045 Received RAS Message = {
23:02:59:045 gatekeeperReject = {
23:02:59:045 requestSeqNum = {
23:02:59:045 1
23:02:59:046 }
23:02:59:046 protocolIdentifier = {
23:02:59:046 {
23:02:59:046 0 0 8 2250 0 5 }
23:02:59:046 }
23:02:59:047 rejectReason = {
23:02:59:047 resourceUnavailable = {
23:02:59:048 NULL
23:02:59:048 }
23:02:59:048 }
23:02:59:048 }
23:02:59:048 }
23:02:59:048
Gatekeeper Reject (GRJ) message received
23:02:59:048 Deleted GRQ Timer.
23:02:59:048 Error: Gatekeeper Reject - Resource Unavailable
23:02:59:049 Error: Gatekeeper error. Either Gk not responding or Gk
sending invalid messages
23:02:59:049 Error: Gatekeeper error detected. Closing GkClient as Gk mode
is UseSpecifcGatekeeper
23:02:59:049 Destroying Gatekeep
--
_
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Re: [asterisk-users] 00h323 cant get gatekeeper to connect
I don't have access to the Avaya (client controlled) - but I will ask for
their config. Does the Avaya allow dumping the config to a file? (Or
screen shots only?)
Here's my ooh323.conf (IP's changed)
Thanks,
MD
[general]
port=1720
bindaddr=99.197.126.160
faststart=yes
h245tunneling=yes
gatekeeper=99.197.126.12
h323id=ObjSysAsterisk
callerid=asterisk
gateway=yes
progress_setup = 8
progress_alert = 8
logfile=/logs/ooh323
[Avaya]
type=peer
context=entryBMW
host=99.197.126.29
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=1720
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Benngård
Sent: Wednesday, March 10, 2010 12:18 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] 00h323 cant get gatekeeper to connect
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without any
problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis mdup...@ocg.ca wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
23:02:59:045 GkClient Received RAS Message
23:02:59:045 Received RAS Message = {
23:02:59:045 gatekeeperReject = {
23:02:59:045requestSeqNum = {
23:02:59:045 1
23:02:59:046}
23:02:59:046protocolIdentifier = {
23:02:59:046 {
23:02:59:046 0 0 8 2250 0 5 }
23:02:59:046}
23:02:59:047rejectReason = {
23:02:59:047 resourceUnavailable = {
23:02:59:048 NULL
23:02:59:048 }
23:02:59:048}
23:02:59:048 }
23:02:59:048 }
23:02:59:048 Gatekeeper Reject (GRJ) message received
23:02:59:048 Deleted GRQ Timer.
23:02:59:048 Error: Gatekeeper Reject - Resource Unavailable
23:02:59:049 Error: Gatekeeper error. Either Gk not responding or Gk
sending invalid messages
23:02:59:049 Error: Gatekeeper error detected. Closing GkClient as Gk mode
is UseSpecifcGatekeeper
23:02:59:049 Destroying Gatekeep
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] BLF and realtime SIP buddies
Hello list,
Can I do something like this for BLF functionality :
[test-blf]
exten = _XX,hint,Macro(GetSIPaccount,${EXTEN})
exten = _XX,hint,SIP/${SIPACCOUNT}
GetSIPaccount is a macro that looks at a MySQL-database for the realtime
table sip_buddies where the SIPusername is taken from.
It works great for internal calls... but how about BLF-functionality ??
Feedback is appreciated !
Thank you,
Jonas.
--
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Re: [asterisk-users] Dialplan behaviour
Yes, this work, thanks! On Wed, Mar 10, 2010 at 11:33 AM, Danny Nicholas da...@debsinc.com wrote: This may just be my opinion, but EG logic works best in an established call, like this [TRONCAL-SIP] exten = 225,1,answer exten=225/91,2,Answer exten=225/91,3,Echo exten=225/92,2,Answer exten=225/92,3,Playback(conf-invalid) exten=225,hangup This way, 225 is answered and hungup regardless of caller, and 91/92 get their specific handling. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, March 10, 2010 8:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Dialplan behaviour I understand the “ex-girlfriend” situation, in fact I want to do that, the problem is when I don´t put the last line and call from 92 or 91 this don´t work. I put the ex-girlfriend exception because without this, calls from 91 and 92 don´t match their extensions. On Mon, Mar 8, 2010 at 4:52 PM, Danny Nicholas da...@debsinc.com wrote: You made the troncal-sip context into a “ex-girlfriend” situation where only calls from extensions 91 and 92 would process. When you added the last line, that made it an open context with two egf exceptions. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, March 08, 2010 1:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dialplan behaviour I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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] func odbc and mult iquery
On Wednesday 10 March 2010 02:09:54 voipas wrote: Does asterisk func odbc support multi query? I'm executing stored procedure which returns two tables. With tsql command I can see both tables. But asterisk only shows the first. My database is MSSQL. Yes, but only in 1.6.0 and above. You'll need to set mode=multirow in func_odbc.conf, and the behavior of func_odbc changes dramatically. See the sample func_odbc.conf for more information. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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] app_confbridge production ready?
On Fri, Mar 5, 2010 at 3:06 PM, Robert McGilvray rmcgi...@globeop.com wrote: Does anyone use confbridge in a large installation and can provide feedback on its stability, quality in comparison to MeetMe? I use a sangoma card in my 1.4.2 box to provide timing and it has never been an issue. Can I expect similar performance from the new timing API? I do not use ConfBridge() in a large installation. I use MeetMe on 1.6.0.* The timing is different for ConfBridge, as it does not require DAHDI. If you have that good of an experience with 1.4, why change anything? -- _ -- 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] 00h323 cant get gatekeeper to connect
hmmm... will be hard to help u without u having access... will do my best. Here is my ooh323.conf anyway... sip:/etc/asterisk# cat ooh323.conf [general] bindaddr=213.88.138.183 port=5088 -- _ -- 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] Extensions.conf changed but not take effect
On Wed, Mar 10, 2010 at 3:38 AM, Zhang Shukun bit...@gmail.com wrote:
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten = _95040X,n(welcome),Background(${welcomefile},,123)
...
exten = i,1,Playback(invalid)
exten = i,2,Goto(${CALLINNUM},welcome)
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
This way you'll capture whatever is matching the _95040X pattern,
instead of the value in the CALLERID(dnid).
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
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Re: [asterisk-users] 00h323 cant get gatekeeper to connect
Are you using a Gatekeeper (CLAN)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Benngård
Sent: Wednesday, March 10, 2010 1:53 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] 00h323 cant get gatekeeper to connect
hmmm... will be hard to help u without u having access... will do my best.
Here is my ooh323.conf anyway...
sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088 --- Port for incoming call defined in signal-group
context=inputinterior.se
dtmfmode=rfc2833
disallow=all
allow=alaw:40 --- Skip :40 and just stay with ulaw
tracelevel=6
[Avaya]
type=friend
context=inputinterior.se
dtmfcodec=127 --- I needed that for proper DTMF signaling
dtmfmode=rfc2833
ip=10.242.14.11 --- IP to my Avaya
port=5088 --- Port for outgoing call defined in signal-group
I don't have access to the Avaya (client controlled) - but I will ask for
their config. Does the Avaya allow dumping the config to a file? (Or
screen shots only?)
Here's my ooh323.conf (IP's changed)
Thanks,
MD
[general]
port=1720
bindaddr=99.197.126.160
faststart=yes
h245tunneling=yes
gatekeeper=99.197.126.12
h323id=ObjSysAsterisk
callerid=asterisk
gateway=yes
progress_setup = 8
progress_alert = 8
logfile=/logs/ooh323
[Avaya]
type=peer
context=entryBMW
host=99.197.126.29
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=1720
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Benngård
Sent: Wednesday, March 10, 2010 12:18 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] 00h323 cant get gatekeeper to connect
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without any
problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
23:02:59:045 GkClient Received RAS Message
23:02:59:045 Received RAS Message = {
23:02:59:045 gatekeeperReject = {
23:02:59:045requestSeqNum = {
23:02:59:045 1
23:02:59:046}
23:02:59:046protocolIdentifier = {
23:02:59:046 {
23:02:59:046 0 0 8 2250 0 5 }
23:02:59:046}
23:02:59:047rejectReason = {
23:02:59:047 resourceUnavailable = {
23:02:59:048 NULL
23:02:59:048 }
23:02:59:048}
23:02:59:048 }
23:02:59:048 }
23:02:59:048 Gatekeeper Reject (GRJ) message received
23:02:59:048 Deleted GRQ Timer.
23:02:59:048 Error: Gatekeeper Reject - Resource Unavailable
23:02:59:049 Error: Gatekeeper error. Either Gk not responding or Gk
sending invalid messages
23:02:59:049 Error: Gatekeeper error detected. Closing GkClient as Gk mode
is UseSpecifcGatekeeper
23:02:59:049 Destroying Gatekeep
--
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Re: [asterisk-users] Extensions.conf changed but not take effect
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
CALLERID(num) instead of CALLINNUM (which is 100% wrong)
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[asterisk-users] PGSQL application
Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 -- _ -- 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] Meetme Closes Conference After One Hour
Hello, I am using asterisk 1.6.2.0 and while running the app_meetme I am finding that all conferences created either statically or in real time are being closed after one hour. This happened even if a set a limit on the conference. Is there I default setting somewhere that I need to adjust? -- _ -- 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] Extensions.conf changed but not take effect
On Wed, Mar 10, 2010 at 2:09 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
CALLERID(num) instead of CALLINNUM (which is 100% wrong)
How is it wrong if he's creating his own variable named CALLINNUM?
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Re: [asterisk-users] PGSQL application
On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote: Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. That application has never been a part of Asterisk in the first place. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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
[asterisk-users] Phishing attempt posing as digium
Did anyone else just get what looks like a phising attempt pretending to be from digium? It appears to be full of links to http://app.en25.com/e/er.aspx I must admit, it looks genuine. -- _ -- 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] Phishing attempt posing as digium
On Mar 10, 2010, at 5:35 PM, Thomas Kenyon wrote: Did anyone else just get what looks like a phising attempt pretending to be from digium? It appears to be full of links to http://app.en25.com/e/er.aspx I must admit, it looks genuine. I suspect you'll find it _IS_ genuine. The en25.com server is an Eloqua box, ... that's the CRM technology Digium uses to track their campaigns. -Darren -- _ -- 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] Asterisk SMDI for Nortel Option 11
This is off topic but I have inherited a Nortel Option 11c and am curious what features you are using that make you want to continue using it in conjunction with Asterisk as opposed to moving completely over to Asterisk. I am just learning the Nortel and find it powerful but haven't found the features that I have become accustomed to in Asterisk. -- _ -- 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] Diaplan reload command not working
I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Extensions.conf changed but not take effect
CALLERID(num) instead of CALLINNUM (which is 100% wrong)
How is it wrong if he's creating his own variable named CALLINNUM?
Maybe you are right, it just looked to me like a typo in for the old and
deprecated ${CALLERIDNUM}.
Philipp
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Re: [asterisk-users] Asterisk SMDI for Nortel Option 11
On Wed, 2010-03-10 at 15:48 -0700, James Noble wrote: This is off topic but I have inherited a Nortel Option 11c and am curious what features you are using that make you want to continue using it in conjunction with Asterisk as opposed to moving completely over to Asterisk. I am just learning the Nortel and find it powerful but haven't found the features that I have become accustomed to in Asterisk. The only thing that prevents me from completely replacing the Nortel thing is the customer. They simply do not want to throw away their investment and spend money on new telephones. The nortel voicemail box failed last year and buying a new one is way more expensive than integrating Asterisk for IVR and voicemail. The only thing they will really miss is the MWI light on their phones. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Asterisk SMDI for Nortel Option 11
Citel makes SIP gateways for Nortel handsets that allow you to forklift the Nortel PBX and drop in Asterisk or any other SIP based platform, while retaining the Nortel desktop experience for the users. Cory J Andrews Carlos Chavez wrote: On Wed, 2010-03-10 at 15:48 -0700, James Noble wrote: This is off topic but I have inherited a Nortel Option 11c and am curious what features you are using that make you want to continue using it in conjunction with Asterisk as opposed to moving completely over to Asterisk. I am just learning the Nortel and find it powerful but haven't found the features that I have become accustomed to in Asterisk. The only thing that prevents me from completely replacing the Nortel thing is the customer. They simply do not want to throw away their "investment" and spend money on new telephones. The nortel voicemail box failed last year and buying a new one is way more expensive than integrating Asterisk for IVR and voicemail. The only thing they will really miss is the MWI light on their phones. -- _ -- 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] PGSQL application
- Tilghman Lesher tles...@digium.com escreveu: On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote: Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. That application has never been a part of Asterisk in the first place. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org Hmm I could swear I used it on the 1.2 days. So, in order to access PostgreSQL directly from the dialplan without the use of AGIs, much like the MYSQL() app, the only way to go is via the function ODBC()? -- _ -- 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] confbridge manager/cli
Jonathan Addleman wrote: However, I can't find any way to interact with an existing confbridge conference. Surely there's some equivalent to meetme's 'meetme list' command? Anything else I can use through the cli or manager API? I just need to list conferences and members. Thanks! Replying to myself - all I've found so far has been to get a list of all the active channels (via manager command CoreShowChannels), and then process all the info. This seems to work for my current small setup, but it's obviously not very efficient, and on a busier system with more active channels, it might really become problematic. Is there a better approach that I'm missing? -- 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] Diaplan reload command not working
On Wednesday 10 March 2010 16:58:58 ayodele abejide wrote: I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Looks like you're missing a letter: dialplan reload -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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] Asterisk SMDI for Nortel Option 11
The only thing they will really miss is the MWI light on their phones. Asterisk supports VMWI Visual MWI in various forms; From a cheap analog phone point of view: have a look at 'mwisendtype' in chan_dahdi.conf line reversal if the phone has a LineReversal LED. neon high voltage pulses, to light NEON bulb. For the fully featured analog phones, that probably have CallerID FSK MWI - the default mwisendtype. SIP phones have their own subscription based VMWI. And I'm sure more. Alec Davis -- _ -- 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] Asterisk SMDI for Nortel Option 11
On Thu, 2010-03-11 at 13:27 +1300, Alec Davis wrote: The only thing they will really miss is the MWI light on their phones. Asterisk supports VMWI Visual MWI in various forms; From a cheap analog phone point of view: have a look at 'mwisendtype' in chan_dahdi.conf line reversal if the phone has a LineReversal LED. neon high voltage pulses, to light NEON bulb. For the fully featured analog phones, that probably have CallerID FSK MWI - the default mwisendtype. SIP phones have their own subscription based VMWI. And I'm sure more. If the phones were directly connected to Asterisk this would not be a problem. All the phones are proprietary Nortel digital phones that connect directly to the Nortel box. I need something like SMDI to signal the PBX to turn the MWI light on. Unfortunately all the documentation I have read says that the Option 11 does not support SMDI and I cannot find any other way of telling it to turn on the light when there is voicemail waiting in Asterisk. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Diaplan reload command not working
A letter, I spelt it right at the CLI prompt but says it does not recognise the command From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Wed, 10 Mar 2010 18:24:51 -0600 Subject: Re: [asterisk-users] Diaplan reload command not working On Wednesday 10 March 2010 16:58:58 ayodele abejide wrote: I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Looks like you're missing a letter: dialplan reload -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Diaplan reload command not working
What version of asterisk are you using? Dialplan reload wasn't added until 1.4. If for some reason you have a 1.2 or older asterisk install, you'll need to use extensions reload (I think, I don't have a 1.2 box in front of me to confirm the exact command). Thanks, --Warren Selby On Mar 10, 2010, at 6:55 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: A letter, I spelt it right at the CLI prompt but says it does not recognise the command From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Wed, 10 Mar 2010 18:24:51 -0600 Subject: Re: [asterisk-users] Diaplan reload command not working On Wednesday 10 March 2010 16:58:58 ayodele abejide wrote: I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Looks like you're missing a letter: dialplan reload -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Si gn up now. -- _ -- 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] Dialplan behaviour
At 4:45 PM on 08 Mar 2010, equis software wrote: I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup [...] Dont work If I add this rule exten=225,1,Answer Works ok I suspect it's because when the call first comes in, asterisk doesn't have the callerid info yet (it comes after the first ring). So asterisk tries to route the call to a callerid-nonspecific dialplan entry, and simply fails when it doesn't find any. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- 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] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
Zoa wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. I tested Attrafax this afternoon and was very pleased to see that it worked first time right out of the box. I tested the gateway function with the Asterisk source in the tarbal, Zaptel 1.4.10.1 and a Digium TE405P ver2 4-port T1 card. I really like the console output while processing faxes. Very impressive. Would anyone mind sharing any performance statistics based on real word usage or even high volume lab testing? I'm wondering how many concurrent T38 to PRI faxes could be handled with high end server hardware. Where are the bottlenecks for the software stack, RAM, PCI Bus, Proc Speed, Disc I/O? Would there be a problem running 3 to 4 PRI's full of T38 to SIP Faxes on one server? Could the Attrafax software handle that volume? Thanks in advanced for any feedback. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Diaplan reload command not working
I am using the 1.6.2.0 version From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Date: Wed, 10 Mar 2010 19:29:02 -0600 Subject: Re: [asterisk-users] Diaplan reload command not working What version of asterisk are you using? Dialplan reload wasn't added until 1.4. If for some reason you have a 1.2 or older asterisk install, you'll need to use extensions reload (I think, I don't have a 1.2 box in front of me to confirm the exact command). Thanks,--Warren Selby On Mar 10, 2010, at 6:55 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: A letter, I spelt it right at the CLI prompt but says it does not recognise the command From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Wed, 10 Mar 2010 18:24:51 -0600 Subject: Re: [asterisk-users] Diaplan reload command not working On Wednesday 10 March 2010 16:58:58 ayodele abejide wrote: I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Looks like you're missing a letter: dialplan reload -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. -- _ -- 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 _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Unable to forward voice or dtmf
Hi all, I am worried because on my production asterisk servers, I am receiving these errors every 2-3 minutes. my log files are full of them: WARNING[xxx] app_dial.c: Unable to forward voice or dtmf and also, less frequent: WARNING[xxx] app_dial.c: Unable to write frame How can I find out what is causing this problem? If anybody can point me in the right direction I would be very grateful. My server is on a public IP address, and all of my customers are behind NAT. I have set canreinvite=no to keep RTP traffic and thus make it easier to get around NAT issues. All of these calls are terminated at another asterisk server with E1 connections, so there is only one place where NAT is present in our network. Also, all of the calls are inbound (NAT'ed customers call the public IP asterisk server) Regards, Alex -- _ -- 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] Phones won't stop ringing
We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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] Diaplan reload command not working
On Wed, Mar 10, 2010 at 8:02 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I am using the 1.6.2.0 version Could you copy and paste your CLI from when you type it in and including the error message? From the cli, you should be able to hit your tab key to see a list of all available commands you can enter. You may need to do that and copy and paste the output from that as well... -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Phones won't stop ringing
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Phones won't stop ringing
Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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] Phones won't stop ringing
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote: On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- 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] Phones won't stop ringing
I'm experiencing runaway ringing too, can we make this a class action against someone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Brower Sent: Wednesday, March 10, 2010 10:20 PM To: Chris Owen Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Phones won't stop ringing Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] Phones won't stop ringing
On Mar 10, 2010, at 10:27 PM, Chris Owen wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i I've been seeing this lately on Cisco 7940, seems to happen on two of the three at a location I deal with. They worked fine for years and then all of a sudden this just started happening. Rebooting the phone will cure it for a period of time, but it always comes back, and always to the same two phones (although not always at the same time). I don't think anything changed when it started happening, but I can't say for sure. It may also happen on a Polycom at that location as well, reports on that one have been sketchy, so I can't be sure it really is versus they are hearing a 2nd call ringing and just think the phone is stuck ringing. (I do know for a fact it happens with the Cisco and is not simply a 2nd call). I had figured it was the old version of Asterisk I'm running and the fact that the server has had several power failures so who knows the health of the machine and install. But if it is happening to others, my assumption may be wrong. -chris www.mythtech.net -- _ -- 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] How to add custom CDR fields to MySQL
Hi all,
I've been trying to add a custom mysql field to my CDR's, but I must
be doing something wrong.
I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add:
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
This extension is executed, I can see it in the asterisk console.
I have added a new column in my MySQL database called q931. However,
the new field does not show up in my database or in the Master.csv
file.
Any help would be greatly appreciated.
Regards,
Alex
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