[asterisk-users] Async Agi problem
Now that everything seems to rock I've hit the next hurdle. In my extensions.conf I have the extension: [agi-async] exten = _01,1,Agi(agi:async) and I can see that the context is hit when dialing into *. However my java app that's supposed to receive async agi events get no such events at all, but it does receive other manager API events. * version is 1.6.1.4 Ideas? /Rob ___ -- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
Simply, You can use Originate command like originate SIP/151 application Meetme 1234,dcs if you want to dial multiple extension then just use while loop . regards Dhaval On Wed, Oct 28, 2009 at 6:45 PM, Danny Nicholas da...@debsinc.com wrote: Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I hadn’t stumbled upon the Async parameter. A “more correct” dissertation of the sentence would be “The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true”. Guess I’ll never be as smart as you, Matt. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Wednesday, October 28, 2009 5:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Hi Matt, That is exactly what I am doing now and it has solved my problem. Now all the calls originate instantly with no noticeable delay. -- Zeeshan A Zakaria On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.com wrote: On 28/10/09 3:52 AM, Danny Nicholas wrote: This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the “first pickup” problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: CDR(billsec)
By setting it I was then able to get the value of the variable in the
script. And all of this happens immediately after the call is hung up
anyway...
Danny Nicholas wrote:
Something seems to be missing here- you don't pass ${BILLTIME} to hangup.php
(as far as I can see), so it seems that hangup.php is operating (at least
somewhat) independently of the dialplan. The OP seemed to want in-line
knowledge of his billable seconds.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Wednesday, October 28, 2009 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] R: CDR(billsec)
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean it’s a bug in 1.4 or an enhancement in 1.6? If the
latter, can the change be back-ported?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Alexandru Oniciuc
*Sent:* Wednesday, October 28, 2009 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] R: CDR(billsec)
Hello Anahi,
I’ve encountered issues with CDR function when I was using the 1.4
version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Wednesday, October 28, 2009 6:35 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0...
but if after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can
I get the call duration in the h extension?
Thanks,
**
**
**Anahi Ludueña**
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
___
-- 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
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
-- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
Dhaval, is this a suggestion or you just copied what I said earlier? -- Zeeshan A Zakaria On Thu, Oct 29, 2009 at 5:15 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Simply, You can use Originate command like originate SIP/151 application Meetme 1234,dcs if you want to dial multiple extension then just use while loop . regards Dhaval On Wed, Oct 28, 2009 at 6:45 PM, Danny Nicholas da...@debsinc.com wrote: Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I hadn’t stumbled upon the Async parameter. A “more correct” dissertation of the sentence would be “The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true”. Guess I’ll never be as smart as you, Matt. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Wednesday, October 28, 2009 5:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Hi Matt, That is exactly what I am doing now and it has solved my problem. Now all the calls originate instantly with no noticeable delay. -- Zeeshan A Zakaria On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.com wrote: On 28/10/09 3:52 AM, Danny Nicholas wrote: This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the “first pickup” problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?
i didnt read above all the mail On Thu, Oct 29, 2009 at 3:06 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Dhaval, is this a suggestion or you just copied what I said earlier? -- Zeeshan A Zakaria On Thu, Oct 29, 2009 at 5:15 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Simply, You can use Originate command like originate SIP/151 application Meetme 1234,dcs if you want to dial multiple extension then just use while loop . regards Dhaval On Wed, Oct 28, 2009 at 6:45 PM, Danny Nicholas da...@debsinc.comwrote: Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I hadn’t stumbled upon the Async parameter. A “more correct” dissertation of the sentence would be “The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true”. Guess I’ll never be as smart as you, Matt. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Wednesday, October 28, 2009 5:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Hi Matt, That is exactly what I am doing now and it has solved my problem. Now all the calls originate instantly with no noticeable delay. -- Zeeshan A Zakaria On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.com wrote: On 28/10/09 3:52 AM, Danny Nicholas wrote: This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the “first pickup” problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Zap inbound hangup problem
Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten = 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from '204263847' to '207582401' on channel 0/31, span 2 -- Executing [207582...@from-pstn:1] Dial(Zap/62-1, Zap/65|30) in new stack -- Called 65 -- Zap/65-1 is ringing -- Zap/65-1 is ringing [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:1797 dahdi_train_ec: No echo training requested [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:4440 dahdi_handle_event: channel 65 answered -- Zap/65-1 answered Zap/62-1 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3662 dahdi_bridge: master: 62, slave: 65, nothingok: 0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3677 dahdi_bridge: Stopping tones on 62/0 talking to 65/0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3689 dahdi_bridge: Stopping tones on 65/0 talking to 62/0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3497 dahdi_link: Making 65 slave to master 62 at 0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:1601 conf_add: Added 77 to conference 3848/1023 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:1601 conf_add: Added 74 to conference 3848/1023 -- Native bridging Zap/62-1 and Zap/65-1 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:3441 dahdi_unlink: Unlinking slave 65 from 62 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:1630 conf_del: Removed 77 from conference 3848/1023 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:1630 conf_del: Removed 74 from conference 3848/1023 -- Executing [...@from-pstn:1] DeadAGI(Zap/62-1, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-5-2) in new stack [Oct 29 11:44:47] WARNING[12424]: res_agi.c:2203 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-5-2completed, returning 0 -- Hungup 'Zap/65-1' == Spawn extension (from-pstn, 207582401, 1) exited non-zero on 'Zap/62-1' [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:3364 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on Zap/62-1 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:2994 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:3360 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/62-1 -- Hungup 'Zap/62-1' [Oct 29 11:44:51] DEBUG[2627]: chan_dahdi.c:7338 do_monitor: Message status for 4065 changed from -1 to 0 on 65 ___ -- 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] Zap inbound hangup problem
Hi On Thu, Oct 29, 2009 at 11:49:17AM +0100, Robin wrote: Hi all, I have an Astribank connected to Asterisk 1.4. DAHDI? Zaptel? What version, exatly? I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? Is the issue reproducable? If so: Could you please increase debug level to 5 and provide a trace from that? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?
On 29/10/09 2:15 AM, Danny Nicholas wrote: Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I hadn’t stumbled upon the Async parameter. A “more correct” dissertation of the sentence would be “The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true”. Guess I’ll never be as smart as you, Matt. :D I should hope not!! If everyone was as smart as me, how would I take over the world? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] R: R: CDR(billsec)
On 29/10/09 5:56 AM, Alexandru Oniciuc wrote:
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Make sure you have endbeforehexten set to yes in /etc/asterisk/cdr.conf
; Normally, CDR's are not closed out until after all extensions are
; finished executing. By enabling this option, the CDR will be ended
; before executing the h extension so that CDR values such as end
; and billsec may be retrieved inside of of this extension.
endbeforehexten=yes
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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] Zap inbound hangup problem
Hi Tzafrir, Ok, here we go... feeling like a dumbass. Your Is the issue reproducable? triggered me. Tried 5 other extensions, no problems on those. So figured it had to be the port or the phone (or some crazy config issue). Tried the phone on another port: same problem. Changed the batteries in the phone: problem solved... sjees! I really have to get my debugging skills back in order here! I think it's the lack of sleep lately trying to get my setup in order ;) Thanks for your response though. At least now I know where to set the debug level... Cheers, Robin On Thu, Oct 29, 2009 at 12:03, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: Hi On Thu, Oct 29, 2009 at 11:49:17AM +0100, Robin wrote: Hi all, I have an Astribank connected to Asterisk 1.4. DAHDI? Zaptel? What version, exatly? I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? Is the issue reproducable? If so: Could you please increase debug level to 5 and provide a trace from that? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- 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] Dynamic DNS trunk
On 29/10/09 2:05 PM, B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. The reply regarding getting the other end to register is correct. The other host should be registering regularly. If you have a host name use /etc/asterisk/dnsmgr.conf: [general] enable=yes ; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] R: R: R: CDR(billsec)
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matt Riddell
Inviato: giovedì 29 ottobre 2009 12.44
A: asterisk-users@lists.digium.com
Oggetto: Re: [asterisk-users] R: R: CDR(billsec)
On 29/10/09 5:56 AM, Alexandru Oniciuc wrote:
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Make sure you have endbeforehexten set to yes in /etc/asterisk/cdr.conf
; Normally, CDR's are not closed out until after all extensions are
; finished executing. By enabling this option, the CDR will be ended
; before executing the h extension so that CDR values such as end
; and billsec may be retrieved inside of of this extension.
endbeforehexten=yes
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote: Thank you! My bad,the CDR function was working on 1.4, I can confirm that endbeforehexten=yes does the trick, I've just tried it :] WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference! :D Yeah based in New Zealand - we're just about ahead of everybody - in fact it's 1:20 in the morning so I probably should go to sleep :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] R: R: R: CDR(billsec)
Thanks Matt! It works now! Bye... Anahi Ludueña Date: Fri, 30 Oct 2009 01:20:02 +1300 From: li...@venturevoip.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] R: R: R: CDR(billsec) On 30/10/09 1:10 AM, Alexandru Oniciuc wrote: Thank you! My bad,the CDR function was working on 1.4, I can confirm that endbeforehexten=yes does the trick, I've just tried it :] WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference! :D Yeah based in New Zealand - we're just about ahead of everybody - in fact it's 1:20 in the morning so I probably should go to sleep :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 _ ¿Sabías que ahora puedes hablar por Messenger desde Hotmail con todos tus contactos? Revisa tu correo mientras conversas con tus amigos. http://www.hotmail.com ___ -- 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] R: R: R: CDR(billsec)
While I dont question that this works, Ill state that for some reason it
doesnt work on my 1.4.26.2 on Centos 5.3
Heres my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
heres cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
heres CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in new
stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3704ae0,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time
0) in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur 0)
in new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in
new stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller hung
up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, Im only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDRs?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, October 29, 2009 7:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
_
Anahi Ludueña
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can confirm
that endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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
_
¿Para qué descargarte juegos, si tienes los más divertidos online? Entra ya
en http://juegosonline.es.msn.com/ Juegos y prepárate para muchas horas
de diversión
___
-- 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] R: R: R: CDR(billsec)
You have to do it in the 'h' exten
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Hangup
exten = h,1,Verbose(time ${CDR(billsec)})
exten = h,n,Verbose(dur ${CDR(duration)})
exten = h,n,Verbose(id ${CDR(uniqueid)})
Ish
Danny Nicholas wrote:
While I don’t question that this works, I’ll state that for some
reason it doesn’t work on my 1.4.26.2 on Centos 5.3
Here’s my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
here’s cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no ; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
here’s CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in
new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4]
WaitMusicOnHold(SIP/170-b3704ae0, 5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time
0) in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur
0) in new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in
new stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller
hung up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in
new stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, I’m only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDR’s?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Thursday, October 29, 2009 7:30 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
**
**
**Anahi Ludueña**
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can
confirm that endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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
¿Para qué descargarte juegos, si tienes los más divertidos online?
Entra ya en Juegos y prepárate para muchas horas de diversión
http://juegosonline.es.msn.com/
Re: [asterisk-users] R: R: R: CDR(billsec)
Good Show, Ish;
This is what I had
exten = s,1,Answer()
exten = s,n,Verbose( caller ${EXTEN})
exten = s,n,Goto(default|s|1)
exten = s,n,Background(vm-goodbye)
exten = s,n,Hangup()
exten = s-NOANSWER,1,Background(vm-goodbye)
exten = s-HANGUP,1,Set(CDR(userfield)=${CDR(userfield)}
Hangupcause:${HANGUPCAUSE})
exten = s-HANGUP,n,AGI(userfield.agi|${CDR(uniqueid)}|${CDR(userfield)})
exten = s-HANGUP,n,Verbose(time ${CDR(billsec)}|${CDR(uniqueid)})
exten = s-HANGUP,n,Hangup(${HANGUPCAUSE})
I Added this
exten = h,1,Set(CDR(userfield)=${CDR(userfield)}
Hangupcause:${HANGUPCAUSE})
exten = h,n,DeadAGI(userfield.agi|${CDR(uniqueid)}|${CDR(userfield)})
exten = h,n,Verbose(time ${CDR(billsec)}|${CDR(uniqueid)})
exten = h,n,Hangup(${HANGUPCAUSE})
and now CLI looks like this
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3706910, ) in new
stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3706910,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3706910,
pls-hold-while-try) in new stack
-- SIP/170-b3706910 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3706910,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3706910
-- Stopped music on hold on SIP/170-b3706910
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3706910,
vm-goodbye) in new stack
-- SIP/170-b3706910 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3706910, time
0) in new stack
time 0oned*CLI
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3706910, dur 0)
in new stack
dur 0honed*CLI
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3706910, id
1256824883.8) in new stack
id 1256824883.8
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3706910, ) in
new stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3706910'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3706910,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3706910,
userfield.agi|1256824883.8| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] Verbose(SIP/170-b3706910, time
9|1256824883.8) in new stack
[Oct 29 09:01:32] WARNING[13921]: app_verbose.c:70 verbose_exec:
'1256824883.8' is not a verboser number
1256824883.8LI
-- Executing [...@dlpn_dialplan1:4] Hangup(SIP/170-b3706910, 16) in
new stack
== Spawn extension (DLPN_DialPlan1, h, 4) exited non-zero on
'SIP/170-b3706910'
So s-HANGUP is not the same as h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, October 29, 2009 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
You have to do it in the 'h' exten
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Hangup
exten = h,1,Verbose(time ${CDR(billsec)})
exten = h,n,Verbose(dur ${CDR(duration)})
exten = h,n,Verbose(id ${CDR(uniqueid)})
Ish
Danny Nicholas wrote:
While I dont question that this works, Ill state that for some
reason it doesnt work on my 1.4.26.2 on Centos 5.3
Heres my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
heres cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no ; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
heres CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in
new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4]
WaitMusicOnHold(SIP/170-b3704ae0, 5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time
0) in new stack
time 0
-- Executing [...@dlpn_dialplan1:7]
Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?
Matt Riddell wrote: snip Guess I’ll never be as smart as you, Matt. :D I should hope not!! If everyone was as smart as me, how would I take over the world? If you really are that smart, why would you want to? Peg Leg O'Brien -- Dog is my co-pilot ___ -- 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] R: R: R: R: CDR(billsec)
Hi Danny,
I'm using CSV output too. Maybe you didn't module reload cdr after adding
endbeforehexten?
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: giovedì 29 ottobre 2009 14.25
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: R: R: CDR(billsec)
While I don't question that this works, I'll state that for some reason it
doesn't work on my 1.4.26.2 on Centos 5.3
Here's my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
here's cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
here's CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3704ae0,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time 0)
in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur 0) in
new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller hung
up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, I'm only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDR's?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, October 29, 2009 7:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
Anahi Ludueña
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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
¿Para qué descargarte juegos, si tienes los más divertidos online? Entra ya en
Juegos y prepárate para muchas horas de
diversiónhttp://juegosonline.es.msn.com/
___
-- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
Matt is smart, but he's probably not a Ivy-League politician :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, October 29, 2009 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Matt Riddell wrote: snip Guess I'll never be as smart as you, Matt. :D I should hope not!! If everyone was as smart as me, how would I take over the world? If you really are that smart, why would you want to? Peg Leg O'Brien -- Dog is my co-pilot ___ -- 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
[asterisk-users] R: R: R: R: R: CDR(billsec)
Ye, don't mind that one ...
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Alexandru Oniciuc
Inviato: giovedì 29 ottobre 2009 15.15
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: [asterisk-users] R: R: R: R: CDR(billsec)
Hi Danny,
I'm using CSV output too. Maybe you didn't module reload cdr after adding
endbeforehexten?
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: giovedì 29 ottobre 2009 14.25
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: R: R: CDR(billsec)
While I don't question that this works, I'll state that for some reason it
doesn't work on my 1.4.26.2 on Centos 5.3
Here's my test snippet
exten = 333,1,answer
exten = 333,n,SetMusicOnHold(default)
exten = 333,n,Background(pls-hold-while-try)
exten = 333,n,WaitMusicOnHold(5)
exten = 333,n,Background(vm-goodbye)
exten = 333,n,Verbose(time ${CDR(billsec)})
exten = 333,n,Verbose(dur ${CDR(duration)})
exten = 333,n,Verbose(id ${CDR(uniqueid)})
exten = 333,n,Hangup
here's cdr.conf
[general]
endbeforehexten = yes
[csv]
usegmtime = no; log date/time in GMT. Default is no
loguniqueid = yes ; log uniqueid. Default is no
loguserfield = yes ; log user field. Default is no
here's CLI output
-- Executing [...@dlpn_dialplan1:1] Answer(SIP/170-b3704ae0, ) in new stack
-- Executing [...@dlpn_dialplan1:2] SetMusicOnHold(SIP/170-b3704ae0,
default) in new stack
-- Executing [...@dlpn_dialplan1:3] BackGround(SIP/170-b3704ae0,
pls-hold-while-try) in new stack
-- SIP/170-b3704ae0 Playing 'pls-hold-while-try' (language 'en')
-- Executing [...@dlpn_dialplan1:4] WaitMusicOnHold(SIP/170-b3704ae0,
5) in new stack
-- Started music on hold, class 'default', on SIP/170-b3704ae0
-- Stopped music on hold on SIP/170-b3704ae0
-- Executing [...@dlpn_dialplan1:5] BackGround(SIP/170-b3704ae0,
vm-goodbye) in new stack
-- SIP/170-b3704ae0 Playing 'vm-goodbye' (language 'en')
-- Executing [...@dlpn_dialplan1:6] Verbose(SIP/170-b3704ae0, time 0)
in new stack
time 0
-- Executing [...@dlpn_dialplan1:7] Verbose(SIP/170-b3704ae0, dur 0) in
new stack
dur 0
-- Executing [...@dlpn_dialplan1:8] Verbose(SIP/170-b3704ae0, id
1256822108.6) in new stack
id 1256822108.6
-- Executing [...@dlpn_dialplan1:9] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, 333, 9) exited non-zero on
'SIP/170-b3704ae0'
-- Executing [...@dlpn_dialplan1:1] Set(SIP/170-b3704ae0,
CDR(userfield)= Hangupcause:16) in new stack
-- Executing [...@dlpn_dialplan1:2] DeadAGI(SIP/170-b3704ae0,
userfield.agi|1256822108.6| Hangupcause:16) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/userfield.agi
-- AGI Script userfield.agi completed, returning 0
-- Executing [...@dlpn_dialplan1:3] NoOp(SIP/170-b3704ae0, id
1256822108.6 time 8) in new stack
-- Executing [...@dlpn_dialplan1:4] NoOp(SIP/170-b3704ae0, caller hung
up) in new stack
-- Executing [...@dlpn_dialplan1:5] Hangup(SIP/170-b3704ae0, ) in new
stack
== Spawn extension (DLPN_DialPlan1, h, 5) exited non-zero on
'SIP/170-b3704ae0'
FWIW, I'm only using the csv CDR; perhaps these values are better
preserved/presented if you use the SQL CDR's?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, October 29, 2009 7:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt!
It works now!
Bye...
Anahi Ludueña
Date: Fri, 30 Oct 2009 01:20:02 +1300
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] R: R: R: CDR(billsec)
On 30/10/09 1:10 AM, Alexandru Oniciuc wrote:
Thank you! My bad,the CDR function was working on 1.4, I can confirm that
endbeforehexten=yes does the trick, I've just tried it :]
WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference!
:D Yeah based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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
¿Para qué descargarte juegos, si tienes los más divertidos online? Entra ya en
Juegos y prepárate para muchas horas de
Re: [asterisk-users] GUI for hunt groups?
www.voiceroute.org also has an open source unified communications manager (they also have a commercial version)... Very little support from the developers but I have deployed it in a few large call centers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, October 28, 2009 7:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] GUI for hunt groups? Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote: On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote: I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons 1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except callerid are recorded properly after every call. I have both a clid and callerid field in the database but both fields are empty. In cdr_mysql.conf I have this alias in the [columns] section: alias start = calldate alias callerid = clid Get rid of this alias callerid = clid line. What it does is to tell the driver to put the CDR variable called callerid into the clid column in the database, overriding the builtin clid mapping. Then reload. If you want the Caller*ID information in the callerid column, then your mapping is backwards and should be alias clid = callerid. Remember, the arrow points in the direction that the information flows: FROM the cdr TO the database. I already tried that with the same result. I even added a callerid column to my cdr table just in case. Either removing the alias line or reversing it like you suggested will not record the callerid in either column. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote: On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote: On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote: I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons 1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except callerid are recorded properly after every call. I have both a clid and callerid field in the database but both fields are empty. In cdr_mysql.conf I have this alias in the [columns] section: alias start = calldate alias callerid = clid Get rid of this alias callerid = clid line. What it does is to tell the driver to put the CDR variable called callerid into the clid column in the database, overriding the builtin clid mapping. Then reload. If you want the Caller*ID information in the callerid column, then your mapping is backwards and should be alias clid = callerid. Remember, the arrow points in the direction that the information flows: FROM the cdr TO the database. I already tried that with the same result. I even added a callerid column to my cdr table just in case. Either removing the alias line or reversing it like you suggested will not record the callerid in either column. Try the following commands. What is output? CLI core set debug 1 CLI module reload cdr_addon_mysql.so -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote: On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote: On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote: On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote: I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons 1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except callerid are recorded properly after every call. I have both a clid and callerid field in the database but both fields are empty. In cdr_mysql.conf I have this alias in the [columns] section: alias start = calldate alias callerid = clid Get rid of this alias callerid = clid line. What it does is to tell the driver to put the CDR variable called callerid into the clid column in the database, overriding the builtin clid mapping. Then reload. If you want the Caller*ID information in the callerid column, then your mapping is backwards and should be alias clid = callerid. Remember, the arrow points in the direction that the information flows: FROM the cdr TO the database. I already tried that with the same result. I even added a callerid column to my cdr table just in case. Either removing the alias line or reversing it like you suggested will not record the callerid in either column. Try the following commands. What is output? CLI core set debug 1 CLI module reload cdr_addon_mysql.so Just this: pbxoficina*CLI core set debug 1 Core debug is at least 1 pbxoficina*CLI module reload cdr_addon_mysql.so -- Reloading module 'cdr_addon_mysql.so' (MySQL CDR Backend) pbxoficina*CLI -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS trunk
Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? If no other way, can I type sip reload on a production system safely? If yes, any help shows how to send the command sip reload periodically to asterisk? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Thursday, October 29, 2009 6:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dynamic DNS trunk If the trunk is a dynamic IP you need the other end to register to Asterisk, so letting Asterisk know the new IP. Regards, Juan B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. _ ___ -- 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] need a local tech
i don't know what your are talking about (sig) B) what trash? c) dont thinks so From: h...@a-domani.nl To: asterisk-users@lists.digium.com Date: Wed, 28 Oct 2009 22:16:16 +0100 Subject: Re: [asterisk-users] need a local tech On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote: I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can figure out a way to talk privately about the details (pay, the problems, etc). If someone knows of a company in the area i am open to that to. __ Windows 7: Simplify your PC. Learn more. ___ -- 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 Hi Ott, Why do you put that URL in your sig? A) This is the non-commercial list B) We rather be refrained from such trash C) Instead of waving a white flag, do a rm -rf / or the M$-equivalent ___ -- 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 _ Windows 7: Simplify your PC. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009___ -- 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] need a local tech
thanks Cohen Date: Wed, 28 Oct 2009 23:46:12 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] need a local tech On Wed, Oct 28, 2009 at 10:16:16PM +0100, Hans Witvliet wrote: On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote: __ Windows 7: Simplify your PC. Learn more. ___ -- 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 Hi Ott, Why do you put that URL in your sig? It's not mim. It's his email provider (the 'windows 7' part) and this list's provider (the rest). He's clearly not advertising himself. So I don't see as issue here. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows 7: Simplify your PC. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009___ -- 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] Async Agi problem
On Thu, Oct 29, 2009 at 3:48 AM, Robert Bielik robert.bie...@xponaut.sewrote: and I can see that the context is hit when dialing into *. However my java app that's supposed to receive async agi events get no such events at all, but it does receive other manager API events. * version is 1.6.1.4 You mean you cannot see AsyncAGI events? did you enable agi in the read= parameter in manager.conf for your Java application user? Can you send AGI commands to the channel through the manager? or through the Asterisk CLI agi exec cmd?? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] Delayed answer when calling out
I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear thank you for calling. Please press a number When placing a call through the LDT1 to the same number. I hear ...Please press a number It is cutting off the Thank you for calling I also notice that I dont hear a ringback... I've tried the following: added r to the dial command (this does give me a ringback but its still cutting off the first few seconds) set usecallerid=no to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] Delayed answer when calling out
Might or might not be relevant. Try dial(DAHDI/X/w#) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Thursday, October 29, 2009 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delayed answer when calling out I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear thank you for calling. Please press a number When placing a call through the LDT1 to the same number. I hear ...Please press a number It is cutting off the Thank you for calling I also notice that I dont hear a ringback... I've tried the following: added r to the dial command (this does give me a ringback but its still cutting off the first few seconds) set usecallerid=no to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] Delayed answer when calling out
Tried it but didn't seem to change anything... You were meaning something like Dial(DAHDI/g2/w1611212,30,tTr) right ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 29, 2009 2:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Delayed answer when calling out Might or might not be relevant. Try dial(DAHDI/X/w#) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Thursday, October 29, 2009 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delayed answer when calling out I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear thank you for calling. Please press a number When placing a call through the LDT1 to the same number. I hear ...Please press a number It is cutting off the Thank you for calling I also notice that I dont hear a ringback... I've tried the following: added r to the dial command (this does give me a ringback but its still cutting off the first few seconds) set usecallerid=no to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] Delayed answer when calling out
Yes. I use POTS here and have to do w# or ww# on some calls. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Thursday, October 29, 2009 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Delayed answer when calling out Tried it but didn't seem to change anything... You were meaning something like Dial(DAHDI/g2/w1611212,30,tTr) right ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 29, 2009 2:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Delayed answer when calling out Might or might not be relevant. Try dial(DAHDI/X/w#) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Thursday, October 29, 2009 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delayed answer when calling out I have a PRI and a LDT1 (em) running... When placing a call through the PRI (to a number with an auto attendant). I hear thank you for calling. Please press a number When placing a call through the LDT1 to the same number. I hear ...Please press a number It is cutting off the Thank you for calling I also notice that I dont hear a ringback... I've tried the following: added r to the dial command (this does give me a ringback but its still cutting off the first few seconds) set usecallerid=no to the LD channel in chan_dahdi.. same results I originally had the T1 set to em_w but it was crashing the server. I set it to em and it seems to stop the crashes... Any thoughts? Thanks, Robert ___ -- 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] !! Unknown IE 50 (cs5, Unknown Information Element) on console.
If you are getting this on your console, and are keen enough to try a patch please have a look at https://issues.asterisk.org/view.php?id=13828 and try the libpri_ie50_cs5-trunk.diff2.txt patch. This IE50 (Codeset 5) is to do with Calling Party Category. https://issues.asterisk.org/view.php?id=13828#112881 If you find it cleans up you console from these messages please report back your success or failure to the mantis bug. Thanks Alec Davis ___ -- 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] Booting Error for /dev/kmem
Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) So this is why, the Asterisk and Zaptel can not start. Any Suggestions Please Thanks a lot Torintino _ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009___ -- 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] Booting Error for /dev/kmem
You've been root-kit'ted. Go into recovery mode and restore your files. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Thursday, October 29, 2009 3:51 PM To: Asterisk-users Subject: [asterisk-users] Booting Error for /dev/kmem Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) So this is why, the Asterisk and Zaptel can not start. Any Suggestions Please Thanks a lot Torintino _ Windows Live: Make it easier for your friends to see what you http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 're up to on Facebook. ___ -- 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] AsteriskForge Now Open
Greetings Asterisk Users, A couple of weeks ago at AstriCon we announced the opening of the AsteriskForge, a collaborative development site for all kinds of Asterisk-related open source projects. The idea behind AsteriskForge is to create a public center for the development and distribution of open source code (both binary and source) that connect with Asterisk. As a challenge to the community, I'm offering a $25 gift certificate from ThinkGeek to the first 10 working software projects (i.e. projects with useful, downloadable, runnable, Asterisk-related software) to move in and set up shop in the Forge. (Sorry - no hardware projects, artwork or poetry qualify for this promotion.) The winners will be picked by the official Digium Prize Selection Committee (me and whomever I can round up in the Digium cafeteria) and all decisions are final. If we have to debate the usefulness of an app, it's probably not useful, so no stuffing the ballot box with Perl AGI scripts that read back the current time. May the forge be with you... Forge QA I've received a few questions about the Forge since the announcement and would like to share the answers with everyone. Please let me know if you have any additional questions. Q) What is the AsteriskForge? A) AsteriskForge is a web site that provides free development and hosting tools for Asterisk-related open source projects. Q) Where is AsteriskForge? A) http://forge.asterisk.org Q) What tools does AsteriskForge include? A) AsteriskForge is built on a platform called GForge Advanced Server. It includes source code control (SVN), file download hosting and tracking (just counts, nothing invasive), mailing lists, forums, documentation, development team management, project management and road-mapping. Q) Who can use AsteriskForge? A) Any open source project with a focus on extending or integrating with Asterisk. The project must be released under an OSI-approved license. Q) What are examples of the kinds of things that will be hosted on AsteriskForge? A) Dialplan and AEL snippets (we have a special section for snippets). AGI scripts and programs in various languages. Desktop tools including screen pop utilities, operator consoles, CRM integration components, and monitoring utilities. Server-side integration tools including unified messaging and collaboration components, scalability and redundancy solutions, web services mash-ups, etc. Administration GUI tools and projects. Prompt packages. Industry-specific (vertical) applications. Power and predictive dialer systems. Pretty much anything that connects with Asterisk. Q) Can I mirror my existing Asterisk-related open source project on AsteriskForge? A) Yes. We are happy to host mirrors of existing Asterisk-related projects. We do require that source downloads be included for each mirrored project, not just binary installers. Q) Is AsteriskForge open to non-software projects? A) Yes, though the tools tend to be fairly software oriented. We are open to hosting documentation, voice prompts, hardware CAD/CAM files, even Asterisk-related poetry if it tickles your fancy. Q) Does Digium get to use AsteriskForge code in commercial products outside of the terms of the selected open source license? A) No. The copyright and commercial rights to the code remain with the author(s). Use of AsteriskForge is not predicated on accepting the Digium Open Source Software Project Submission Agreement. Q) What other rules and regulations should I know about? A) You can't post stuff you don't own. We'll honor take-down notices if they are deemed to be legitimate by our legal staff. No development of proprietary commercial products - you must release your code as open source. We won't host .ISO install images or other huge binaries. The full list of terms and conditions are available here: http://www.asterisk.org/forge/terms Q) Is the core of Asterisk moving into AsteriskForge? A) No. There's already a full set of development tools and processes in place for Asterisk, and moving to the AsteriskForge site would be disruptive to the process. We will continue to use the existing issue tracker, mailing lists, review board, forums and subversion repositories for Asterisk. Thanks, -S Steven Sokol Digium, Inc. | Marketing Director - Asterisk 1568 South Yorktown Place – Tulsa, OK – 74104 direct: +1 256-428-6101 mobile: +1 816-806-8844 fax: +1 816-817-0441 twitter: ssokol | jabber: sso...@digium.com | skype: ssokol.digium ___ -- 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] Booting Error for /dev/kmem
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) On Thu, 29 Oct 2009, Danny Nicholas wrote: You've been root-kit'ted. Go into recovery mode and restore your files. Any time you suspect that a box has been compromised the only solution is to pull the drives, replace them with fresh drives and install from the CD/DVD and your backups. What if the cracker munged your recovery mode to erase the drives or to plant itself back into your recovered system? You cannot trust any executable or script from the old drives. If you need data from the old drives, mount them as non-boot drives, copy the data and then label them as compromised and put them on the shelf until you know you don't need anything from them and then re-format. This assumes you aren't looking to go legal. Then you have to learn about chain of custody and preserving evidence. You should also examine every host on your network as well as any system that trusts this 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astreicon presentations
Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :) ___ -- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
On 29/10/09 22:40, Matt Riddell wrote: :D I should hope not!! If everyone was as smart as me, how would I take over the world? With violence, just like everyone else! PaulH ___ -- 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] need a local tech
Sorry, I'm top-posting because I don't know how to do the indenting thing in Microsoft Outlook (some would say I shouldn't be using that, but that's what I have). Tzafrir says I don't see as issue here, but there is a clear issue. In the snip'd message below, it's not clear who said it. I think it was Hans Witvliet [h...@a-domani.nl]. He asked Why do you put that URL in your sig? (Tzafrir explained this) A) This is the non-commercial list (This concern is not applicable as the poster didn't include the footnote.) B) We rather be refrained from such trash (This is rude.) But, importantly, C) Instead of waving a white flag, do a rm -rf / or the M$-equivalent is something that a poster pleading for help and not familiar enough with Linux or, desperately trying anything, not thinking things through, might do. That's not the kind of collegial relationship that I think members of a list like this should expect. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, October 29, 2009 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] need a local tech thanks Cohen Date: Wed, 28 Oct 2009 23:46:12 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] need a local tech On Wed, Oct 28, 2009 at 10:16:16PM +0100, Hans Witvliet wrote: On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote: __ Windows 7: Simplify your PC. Learn more. ___ -- 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 Hi Ott, Why do you put that URL in your sig? It's not mim. It's his email provider (the 'windows 7' part) and this list's provider (the rest). He's clearly not advertising himself. So I don't see as issue here. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Server with Panasonic PBX
On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: C F thankyou very much. when i make a call to Asterisk server recieves and works fine. But as to make external calls we have to press nine so supposed a logic to dial 9 first then wait and then dail other number. But as i dail 9 asterisk show the call as connected with alot of noise. Please help in how to handle this How are you connected from astersik to the TDA? On a side note, may I ask why you are integrating asterisk with the TDA? What is the functionality you plan on gaining? Nothing very important logical its a client who don't want to trash its existing system. So we need to do that. I know Asterisk is far more better to use and handle his requirements but What requirement? Asterisk is NOT the solution to everything. If fact for some it might create more headaches than you would wish. In any event what exactly is the Asterisk system adding here that Panasonic couldn't handle? On Thu, Oct 29, 2009 at 5:25 AM, C F shma...@gmail.com wrote: Any simple legacy integration will work. Search on voip-info.org Here are some problems that I know exist with panasonic systems on their SLT (analog) ports: 1. No CPC, Asterisk if connected using station ports on the TDA to FXO on asterisk, will not detect hangups since the TDA will not send them. 2. BLF and the like will not work. 3. There are different ways of making sure that asterisk users should be able to use the lines on the TDA depending on how you chose to connect them both. On a side note, may I ask why you are integrating asterisk with the TDA? What is the functionality you plan on gaining? On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that will be Asterisk system. I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic to in of Asterisk PBX). --But i am in doubt if i can make Asterisk to make calls outside from the existing PBX ?(ie usually press nine and then wait for a line. In Asterisk system we will dail 9 first then wait then dail the number). Please share your ideas and experience. All the calls will be recieved by existing Panasonic PBX and an Operator will forward calls to Asterisk PBX ... this is requirement. Please also let me know which type of hardware will be required at Asterisk end to handle lines from a PBX. -- Best Regards Shakeel Abbas ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] Dynamic DNS trunk
On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? Yep /etc/asterisk/dnsmgr.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Unable to set TOS to 184?
I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? Thanks, Bart___ -- 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] Unable to set TOS to 184?
If you already mangle packets with IPTABLES, then you should comment the line[s] tos_* on sip.conf. Regards, Juan Bart Fisher wrote: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? Thanks, Bart ___ -- 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] Dynamic DNS trunk
Thanks I did this dnsmgr.conf: enable=yes refreshinterval=300 I did dnsmgr refresh, the DNS in the trunk did not got the new ip, also I waited 5 min. do I have to add an entry to dnsmgr?? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, October 30, 2009 1:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dynamic DNS trunk On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? Yep /etc/asterisk/dnsmgr.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
Please don't make this email thread a place for political discussion. Reply only if you have any other ideas on how to accomplish what was asked in the beginning. Thanks for your understanding. -- Zeeshan A Zakaria On Thu, Oct 29, 2009 at 5:59 PM, Paul Hales pdha...@optusnet.com.au wrote: On 29/10/09 22:40, Matt Riddell wrote: :D I should hope not!! If everyone was as smart as me, how would I take over the world? With violence, just like everyone else! PaulH ___ -- 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] Unable to set TOS to 184?
On Thu, 2009-10-29 at 16:36 -0700, Bart Fisher wrote: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? snip I'm pretty new to Asterisk so take this with a grain of salt. Is it possible you used decimal (184) instead of hex notation (b8) in your sip.conf? We're running 1.6.1.6 and it appears to be working just fine. Here are the pertinent lines from our sip.conf: tos_audio=0xb0 ; b8 (expedited forwarding) confuses the Linux pfifo_fast so b0 works better for us tos_sip=0xb0 The comment is also important in light of the iptables rules you have. As someone else pointed out, you shouldn't need both. I prefer to set them in the application. For example, if I ever change ports for whatever reason, I won't have the problem of forgetting to also change my iptables rules. Now, I may be wrong about this so I wouldn't mind feedback from someone who know better than I do, but I think expedited forwarding (ef = 184 = b8) can shoot you in the foot in Linux. If you don't change the default packet queueing from pfifo-fast, I believe it will not look at the DSCP bits but rather the ToS bits and will place ef packets into band1 (normal priority) rather than band0 (high priority). That's why we use b0 instead and then tell our DSCP enabled switches to place the resultant DSCP values into the highest priority queue. Hope that makes sense. If I'm wrong, please, someone call me out on it. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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
