[asterisk-users] TTS for asterisk
Hi all, Any good TTS (free or commercial) for asterisk? Rgds, Ringo -- _ -- 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] TTS for asterisk
Thanks. Do it support multi-language? On Mon, May 24, 2010 at 11:55 PM, Steve Edwards asterisk@sedwards.com wrote: On Mon, 24 May 2010, Rilawich Ango wrote: Any good TTS (free or commercial) for asterisk? I like Cepstral with the Allison (Smith) font. Allison Smith does the sounds distributed with Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS for asterisk
Thanks. Actually, I am looking for a TTS that support Chinese (Mandarin and Cantonese). Do you have any suggestion? Up to now, I can't find any TTS can support Chinese. As I know Lumenvox is a voice recognition engine. Is it also a TTS? ango On Tue, May 25, 2010 at 12:51 AM, Danny Nicholas da...@debsinc.com wrote: The Cepstral paid version has several languages available and other voices for those 10 people who don't like Allison. At $35.00 a pop, it's not prohibitive (Lumenvox is much more pricey) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rilawich Ango Sent: Monday, May 24, 2010 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TTS for asterisk Thanks. Do it support multi-language? On Mon, May 24, 2010 at 11:55 PM, Steve Edwards asterisk@sedwards.com wrote: On Mon, 24 May 2010, Rilawich Ango wrote: Any good TTS (free or commercial) for asterisk? I like Cepstral with the Allison (Smith) font. Allison Smith does the sounds distributed with Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] voice recognition suggestion
Hi all, I am looking for a voice recognition technology integrated to asterisk. Any suggestion about it? Ango -- _ -- 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] question about call transfer
Hi all, Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf. It shows we can use variable BLINDTRANSFER to call back the one who transfer the call. However, in my tests below. The result is not as expected. case 1: A calls B (dial(sip/B||Tt) B answers and connects to A B transfer to C C doesn't answer the call and B ring again case 2: A calls B (dial(sip/B||Tt) B answers and connects to A A transfer to C C doesn't answer the call but B ring instead of A In case 2, the person who transfer the call can't get back the call. Anyone can tell whether there is a way to correct in case 2? Thanks, ango ___ -- 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] queues autopause
Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in 25000 ms -- SIP/1234-1544cd90 is ringing Is it the limitation of the asterisk to support one queue of autopause function? Or any setting I need to take care to make autopause function works for all queue? ango ___ -- 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] queues autopause
Thanks. Finally, I find that it was caused by the use of the table wrongly. On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina mmol...@millenium.com.co wrote: Rilawich Ango escribió: Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in 25000 ms -- SIP/1234-1544cd90 is ringing Is it the limitation of the asterisk to support one queue of autopause function? Or any setting I need to take care to make autopause function works for all queue? ango ___ -- 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, Some months ago there was a discussion about this, with a simple solution involving minimal changes to the source (1 line of code). Search the archives of this list and you will find the answer. BTW, what version of asterisk are you using? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] question about getting instance ringing member in queue
Hi, I have a queue and 3 agents in the queue like below SIP/1001 SIP/1002 SIP/1003 When I dial the queue number, the agent start to ring. How can I get the instance ringing agent as I want to pause the agent (pausequeuemember) after the queue timeout? Any application or variable can use to get the ringing agent? ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play prompt after hanup
Thanks. Is it possible to do the same after Queue command? After Queue command, hangup will hangup the call and won't go to the next priority. On Mon, Aug 17, 2009 at 7:22 PM, Trevor Hammondstre...@concipient.net wrote: On Monday, August 17, 2009, Rilawich Ango wrote: Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the dialplan doesn't go to s-ANSWER. -- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150, SIP/3001|50|Tt) in new stack -- Called 3001 -- SIP/3001-0986d1d8 is ringing -- SIP/3001-0986d1d8 answered SIP/10.31.0.32-09872150 == Spawn extension (default, 3001, 12) exited non-zero on 'SIP/10.31.0.32-09872150' You need to ensure you specify the g option when you dial the destination (e.g. Dial(SIP/3001,50,Ttg)). Otherwise the call will jump to the h exten when either party hangs up. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No translator path exists for channel type dahdi
Hi, I have a 4 port analog cards with asterisk 1.4.26.1 (centos5.3) installed. After I dial an outgoing call, it returns error and call drop as below. Anyone can tell me what the problem is. ango -- Executing [8...@internal:20] Dial(SIP/601-09425ab8, dahdi/g0/8200|50|T) in new stack [Aug 25 16:43:40] WARNING[2505]: channel.c:3372 ast_request: No translator path exists for channel type dahdi (native 76) to 256 [Aug 25 16:43:40] WARNING[2505]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'dahdi' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [8...@internal:21] Hangup(SIP/601-09425ab8, ) in new stack == Spawn extension (internal, 8200, 21) exited non-zero on 'SIP/601-09425ab8' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play prompt after hanup
Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the
dialplan doesn't go to s-ANSWER.
-- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150,
SIP/3001|50|Tt) in new stack
-- Called 3001
-- SIP/3001-0986d1d8 is ringing
-- SIP/3001-0986d1d8 answered SIP/10.31.0.32-09872150
== Spawn extension (default, 3001, 12) exited non-zero on
'SIP/10.31.0.32-09872150'
On Mon, Aug 17, 2009 at 3:12 PM, DHAVAL
INDRODIYAdhaval.it01...@gmail.com wrote:
Please Use DIALSTATUS Application variable
exten = s,n,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Noop(please do action in next priority)
exten = s-ANSWER,2,Playback(demo-instruct)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Set(DTIME=$[${EPOCH} - ${DIALSTART}])
exten = s-NOANSWER,2,,Hangup()
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,hangup
exten = s-CONGESTION,1,Congestion
On Mon, Aug 17, 2009 at 8:24 AM, Rilawich Ango maillist...@gmail.com
wrote:
HI,
Actually, I want to do the following.
A (user) talks to B (CS). At the end of the talk, B hangup and A will
goto the survey system. That's why I need to play prompt for the user
after hangup. Is it possible?
On Fri, Aug 14, 2009 at 6:32 PM, Trevor Hammondstre...@concipient.net
wrote:
On Friday, August 14, 2009, Rilawich Ango wrote:
Hi,
Can I play a prompt after hanging up a call? I have tried below but
failed.
...
exten = s,n,Dial(SIP/1234)
...
exten = h,1,Playback(demo-instruct)
-- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0,
demo-instruct) in new stack
[Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback:
Failed to write frame
-- SIP/3601-09856bf0 Playing 'demo-instruct' (language 'en')
Rilawich,
If the channel has been hung up, where do you expect the prompt to be
played?
Sincerely,
Trevor Hammonds
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Re: [asterisk-users] play prompt after hanup
HI, Actually, I want to do the following. A (user) talks to B (CS). At the end of the talk, B hangup and A will goto the survey system. That's why I need to play prompt for the user after hangup. Is it possible? On Fri, Aug 14, 2009 at 6:32 PM, Trevor Hammondstre...@concipient.net wrote: On Friday, August 14, 2009, Rilawich Ango wrote: Hi, Can I play a prompt after hanging up a call? I have tried below but failed. ... exten = s,n,Dial(SIP/1234) ... exten = h,1,Playback(demo-instruct) -- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0, demo-instruct) in new stack [Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback: Failed to write frame -- SIP/3601-09856bf0 Playing 'demo-instruct' (language 'en') Rilawich, If the channel has been hung up, where do you expect the prompt to be played? Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play prompt after hanup
Hi, Can I play a prompt after hanging up a call? I have tried below but failed. ... exten = s,n,Dial(SIP/1234) ... exten = h,1,Playback(demo-instruct) -- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0, demo-instruct) in new stack [Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback: Failed to write frame -- SIP/3601-09856bf0 Playing 'demo-instruct' (language 'en') ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP phone recommendation
Hi all, Any good recommendation of IP phone in term of sound quality and price (reasonable) using with asterisk? ango ___ -- 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] regarding to field of accountcode
Thanks. I wonder do I need to reload it if I am using realtime/database? I have to change the accountcode during the call so it is not possible to do it if reload is needed. On Fri, May 29, 2009 at 9:35 PM, Tarek Sawah tareksa...@hotmail.com wrote: accountcode is a setting you add to your SIP peer.. so it doesn't require restarting Asterisk.. only restart the SIP module.. sip reload will be enough my friend.. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Fri, 29 May 2009 17:21:08 +0800 From: maillist...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] regarding to field of accountcode Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango ___ -- 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 Live™: Keep your life in sync. Check it out. ___ -- 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] regarding to field of accountcode
Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango ___ -- 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] regarding to field of accountcode
I am using 1.4.24 with realtime. On Fri, May 29, 2009 at 5:21 PM, Rilawich Ango maillist...@gmail.com wrote: Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango ___ -- 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] asterisk-addon 1.6.1 problem
Hi all, I download asterisk-addon 1.6.1 but the VoIP phone failed to register to the system with the message below. [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk sip I use the same configuration file (res_mysql.conf extconfig.conf) in 1.6.0 but failed. Any big change in 1.6.1? ___ -- 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-addon 1.6.1 problem
I follow it to set [readhost.asterisk] and [writehost.asterisk] and extconfig.conf sippeers = mysql,readhost.asterisk/writehost.asterisk,sipfriends. However the error message still existed. Can you give me an example of res_mysql.conf and extconfig.conf? On Tue, May 26, 2009 at 10:33 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote: Hi all, I download asterisk-addon 1.6.1 but the VoIP phone failed to register to the system with the message below. [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk sip I use the same configuration file (res_mysql.conf extconfig.conf) in 1.6.0 but failed. Any big change in 1.6.1? Please read UPGRADE.txt in the asterisk-addons directory. -- Tilghman ___ -- 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] interruption in queue
HI, I want to allow user to press 0 to the voicemail if the user don't want to wait in the queue. Below is what I set but it doesn't work. Anyone can help? ango file: features.conf [applicationmap] opervm = 0,self/both,Macro,opervm file: extensions.conf ... exten = ,n(queue),Set(DYNAMIC_FEATURES=opervm) exten = ,n,Queue(|tThH|||180) ... [macro-opervm] exten = s,1,NoOp(--openvm--) exten = s,n,VoiceMail(3...@default,u) exten = s,n,Hangup ___ -- 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] interruption in queue
Thanks all. I figure out to exit the queue by setting context in queue.conf. On Thu, May 21, 2009 at 11:20 PM, Kevin P. Fleming kpflem...@digium.com wrote: Mark Michelson wrote: Not to undermine Kevin's requests to read what is documented, I can say that what you want actually will not be presented by running core show application Queue in the CLI. As file would say... 'osnap' In my haste to respond this morning while eating breakfast I didn't look for where that was actually documented. Sorry :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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 ___ -- 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 avoid call waiting? Or check DIALSTATUS before Dial()?
Can you try to disable call waiting in your phone?
On Fri, May 15, 2009 at 6:44 AM, sean darcy seandar...@gmail.com wrote:
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Here I dial out:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] Set(DAHDI/1-1, CALLERID=house
2127873453) in new stack
-- Executing [...@internal:3] Dial(DAHDI/1-1,
.
And now an incoming call:
-- Executing [...@incoming:1] Answer(IAX2/nhi-10929, ) in new stack
-- Executing [...@incoming:2] Dial(IAX2/nhi-10929, DAHDI/1,60) in
new stack
-- Called 1
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- DAHDI/1-2 is ringing
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- CPE supports Call Waiting Caller*ID. Sending 'Seandarcy/212 573
1432'
Is there a way to check the status of a dahdi channel _before_ dialing it?
exten =s,n,ExecIf($[DAHDI/1${DIALSTATUS} =
BUSY]?Dial(${secondline},30)) ??
What's special control 20 ??
Any help appreciated.
sean
BTW, this is on 1.6.1.
sean
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[asterisk-users] question of flite installation
Hi, After following the messages to install flite, I can find the following files. /usr/lib/asterisk/modules/app_flite.so /etc/asterisk/flite.conf That's mean flite is installed successfully. Then I restart asterisk but nothing found for that module. sip*CLI core show application flite Your application(s) is (are) not registered The module doesn't load by asterisk even after restart. How can Asterisk reload all the modules in /usr/lib/asterisk/module? ango ___ -- 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] music on hold using mms
Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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] music on hold using mms
Thanks. But I heard that mpg123 uses much resources (CPU memory) of each connection. Is it true? How about using madplay? On 4/28/09, M Hulber asterisk-ad...@hulber.com wrote: Didn't do mms but have implemented using Shoutcast. I have instructions at the link below: http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ Rilawich Ango wrote: Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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 -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] function originate
As what you said, it is very difficult to control if meetme is created for each call. Playing a message after party A answers if a choice but party A will still need to hear ring after the message. She may still feel weird. Just want to know the purpose of parameter async. Can anyone tell me how to use and in which situation to use it? On Fri, Apr 24, 2009 at 5:06 PM, Geraint Lee gera...@gmail.com wrote: You could use 2 originate commands and connect both of them to a meetme room? But surely what you're trying to do is going to confuse the person anyway if they don't hear anyone when they answer? Wouldn't it just be better to play a message after party a answers and then start ringing party b so that party a knows what's going on? 2009/4/24 Rilawich Ango maillist...@gmail.com Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call --- party A party A rings party A answers call party B rings, party A still hear ring party B answers and A B connected. party A will feel weird when she will still hear ring after answering a call until party B answers it. Below is what I want to do: originate call --- party A party A rings party B rings party A answers call A B connected. ___ -- 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] function originate
Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call --- party A party A rings party A answers call party B rings, party A still hear ring party B answers and A B connected. party A will feel weird when she will still hear ring after answering a call until party B answers it. Below is what I want to do: originate call --- party A party A rings party B rings party A answers call A B connected. ___ -- 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] voice quality
Normally, there are 10 concurrent calls in peak. You are right that usage g729 is due to bandwidth consideration. On Thu, Apr 23, 2009 at 2:42 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Thu, 23 Apr 2009, Rilawich Ango wrote: Hi all, I wonder who has the same voice quality problem as what we have. Below is our configuration. Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer Sometimes, customers told me that they heard our voice not very clear, like a call from far far away. I heard the recording is ok and there is no such effect in it. Can I assume the following? -voice quality is ok in asterisk as recording is ok -The far far away effect is happen between asterisk and customer end Anyone can give me some suggestions to solve/test it? How many concurrent calls are you making? Not using G729 between the Asterisk box and the Cisco would be a start - at least for 1 or 2 calls - but I guess you'res using g729 due to bandwidth restrictions... Gordon ___ -- 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] voice quality
Hi all, I wonder who has the same voice quality problem as what we have. Below is our configuration. Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer Sometimes, customers told me that they heard our voice not very clear, like a call from far far away. I heard the recording is ok and there is no such effect in it. Can I assume the following? -voice quality is ok in asterisk as recording is ok -The far far away effect is happen between asterisk and customer end Anyone can give me some suggestions to solve/test it? ango ___ -- 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] fail to retrieve the calling party information
HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- 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] fail to retrieve the calling party information
Thanks. Let me try it. On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote: That's because you have to create a user account in sip.conf ... + Asterisk is sensitive about it. What should help is if you register the phone with that sip account first. Martin On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote: HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- 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] conference function problems
The CLI shows zap is necessary for conference recording. Can I enable conference recording if using ztdummy or dahdi, how? ango -- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520, 5599|rcixMP) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '5599' [Mar 31 17:57:39] WARNING[22242]: app_meetme.c:2375 find_conf_realtime: No Zap channel available for conference, user introduction disabled [Mar 31 17:57:39] WARNING[22242]: app_meetme.c:2381 find_conf_realtime: No Zap channel available for conference, conference recording disabled -- SIP/3601-c80b4520 Playing 'conf-getpin' (language 'en') ___ -- 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] hum noise
My configuration is simple as below. SIP phone - asterisk - CISCO - T1 Do you mean the hum noise is created by electric-magnetic field? Asterisk can do nothing to eliminate it? On Sun, Mar 29, 2009 at 2:47 AM, Steve Edwards asterisk@sedwards.com wrote: On Sat, 28 Mar 2009, Rilawich Ango wrote: We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango It depends on the source :) What computer, what type of connection (Zap (I'm a 1.2 Druid), SIP, IAX), what interface hardware (none, t100p, etc.)? Anything else like the channel bank sits atop the isolation transformer for the entire building? I built an Asterisk server out of an old Fiire Station. It's Via micro-atx in a shoe-box and a tdm400. Having everything crammed into such a small space put the horizontally mounted tdm card millimeters above the CPU. The noise during dialtone was so bad I scrapped the project there and then. 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 ___ -- 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] hum noise
HI, We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango ___ -- 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] field lastms in 1.4.24
Hi all, I found that a new field lastms is used in 1.4.24. What is the usage of that field and the datatype of it? ango ___ -- 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] field lastms in 1.4.24
Tilghman, Thanks. Can you elaborate the usage about it? What is the meaning of each valid value in this field? ango On Mon, Mar 23, 2009 at 11:24 AM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote: Hi all, I found that a new field lastms is used in 1.4.24. What is the usage of that field and the datatype of it? It's an integer field used to ensure that realtime qualify continues to function across a reload event. -- Tilghman ___ -- 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] recording (mixmonitor) stopped of transfer/call parking after queue
Hi all, I enabled recording (mixmonitor) in queue and process started after queue member pick the call. But recording will stop after picking up by another extensions of call transfer/parking in the same call. Is it possible to continue to record the call for call parking/transfer, how? Rgds, ango ___ -- 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] multiple asterisks in a server
Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango ___ -- 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] multiple asterisks in a server
It seems better to install once with multiple instances. Do we need to take care the port or IP of each instance? On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Klaus Darilion wrote: Rilawich Ango wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Just to have several Asterisk instances on a single server you do not need to install it multiple times. Install it once and start it multiple times. Of course you have to have a dedicated configuration for each server, eg: /etc/asterisk/instance1/* /etc/asterisk/instance2/* /etc/asterisk/instance3/* Then you start the Asterisk process and specify the location of the asterisk.conf file. asterisk -C /etc/asterisk/instance1/asterisk.conf asterisk -C /etc/asterisk/instance2/asterisk.conf asterisk -C /etc/asterisk/instance3/asterisk.conf Further, in asterisk.conf specify for each asterisk instance a different location of: spool directory, PID file, btw: I use a common /var/lib/asterisk/ as I want to have the same sounds for all instances. This gives a problem when you use 1.4, as 1.4 can not configure the location of astdb. For these you have to apply this patch: http://bugs.digium.com/view.php?id=14257 regards klaus ___ -- 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] multiple asterisks in a server
Can you elaborate more how to use 2 IPs for 2 instances of asterisk? On Tue, Feb 24, 2009 at 5:44 PM, Geraint Lee gera...@gmail.com wrote: Almost forgot, you need to make sure you bind each instance to either it's own IP address or different ports on the same ip, i used 2 IP's for it and never hda a problem. 2009/2/24 Geraint Lee gera...@gmail.com Yes it's possible.. When you install use... ./configure --prefix=/usr/local/asterisk2 or something like it. I had to change astrundir (in asterisk.conf) as well. One thing to watch out for is that if you run make samples it will overwrite the ones stored in /etc/asterisk and not where you'd expect them to be in /usr/local/asterisk2/etc/asterisk (or at least it di dwhen i did it!). and for a helping hand i symlinked /usr/local/asterisk2/sbin/asterisk to /usr/local/sbin/asterisk2 and /usr/local/asterisk2/sbin/safe_asterisk to /usr/local/sbin/safe_asterisk2 Cheers Geraint You will also need to look at asterisk.conf in the new installation directory and as a quickfix to get it running, use a different location for astrundir 2009/2/24 Rilawich Ango maillist...@gmail.com - Show quoted text - Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango ___ -- 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] Broken Pipe error while using UpdateConfig command
I also experience that problem. Is it a bug? On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com wrote: Remco Barendse wrote: 1.4.23.1 is quite badly broken and there are no significant new features There are no new features at all, actually. What problems are you having with 1.4.23.1? It doesn't accomplish much to say that it is quite badly broken without at least telling what is wrong. We can't fix what's wrong if we don't know what's wrong to begin with. :) Mark Michelson ___ -- 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] bridge 2 calls
Thanks all. I think click to call can fulfill my purpose. On Thu, Jan 15, 2009 at 6:10 PM, Dovid Bender asteriskus...@dovid.net wrote: I gues understood his email wrong. Seemed to be that he wante to make 2 calls via the web and bridge them. - Original Message - From: C. Savinovich c.savinov...@itntelecom.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, January 15, 2009 2:46 AM Subject: Re: [asterisk-users] bridge 2 calls None of these examples actually create a 3-way call, which is, unless I am mistaken, the original request. An incoming/outgoing call gets bridged to a local channel alright, but then how do you bridge that call to yet another call?. I did try some alternatives and the only way I found is by using a meeting room. Not too elegant in my opinion although it works nicely. If anyone knows of a better way please tell me. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, January 14, 2009 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bridge 2 calls I use post variables. I found this on the web. Forgot where I got it from (sorry that I can't give you credit). ?php //Connect to the Asterisk Manager $socket = fsockopen(127.0.0.1,5038, $errno, $errstr); fputs($socket, Action: Login\r\n); fputs($socket, UserName: username\r\n); fputs($socket, Secret: password\r\n); fputs($socket, Events: off\r\n\r\n); fputs($socket, \r\n\r\n); fputs($socket, Action: Originate\r\n); fputs($socket, Channel: SIP/.$_POST['first_call'].@my_peer\r\n); fputs($socket, Context: mycontext\r\n); fputs($socket, Exten: .$_POST['local_exten'].\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, Callerid: 5551212\r\n); fputs($socket, Timeout: 10\r\n); fputs($socket, Variable: FOO=.$my_var.\r\n); fputs($socket, \r\n\r\n); fputs($socket, \r\n); fputs($socket, Action: Logoff\r\n\r\n); fclose($socket); ? - Original Message - From: Nick Wolf new...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 06, 2009 12:18 PM Subject: Re: [asterisk-users] bridge 2 calls I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call transfer in CDR
Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango ___ -- 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] bridge 2 calls
HI... I found that there is a cmd called, bridge in 1.6. Does it work what I expect? How can I use it if it works as I can't find any information about it in web? On Tue, Jan 6, 2009 at 6:18 PM, Nick Wolf new...@gmail.com wrote: I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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] bridge 2 calls
Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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] remove queue call
Yup I just copy and paste to it but it shown not a known channel. On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote: Did you tab complete it to make sure it was right? On 28 Aug 2008, at 11:39, Rilawich Ango wrote: I got the message below after I issue the soft hangup. sip01*CLI soft hangup Local/[EMAIL PROTECTED],2 Local/[EMAIL PROTECTED],2 is not a known channel Any other way to kill the call without affecting other queues and calls? On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes [EMAIL PROTECTED] wrote: Try CLI soft hangup Local. On 28 Aug 2008, at 09:01, Rilawich Ango wrote: Hi , Actually, there are 3 queues in the server. Only one queue (2700) has problem. I want to reset or remove the caller only in 2700 without affecting other queues or calls. Does it work for my case? On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove queue call
Hi , Actually, there are 3 queues in the server. Only one queue (2700) has problem. I want to reset or remove the caller only in 2700 without affecting other queues or calls. Does it work for my case? On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove queue call
I got the message below after I issue the soft hangup. sip01*CLI soft hangup Local/[EMAIL PROTECTED],2 Local/[EMAIL PROTECTED],2 is not a known channel Any other way to kill the call without affecting other queues and calls? On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes [EMAIL PROTECTED] wrote: Try CLI soft hangup Local. On 28 Aug 2008, at 09:01, Rilawich Ango wrote: Hi , Actually, there are 3 queues in the server. Only one queue (2700) has problem. I want to reset or remove the caller only in 2700 without affecting other queues or calls. Does it work for my case? On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] remove queue call
Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue member state
I have a realtime queue and the state of the queue member change as below. Not-in-use (no call)- Unknown (ringing)- Not-in-use (answered). The state shown in show queues does not really reflect the state of the phone. I have searched the net and also the UPGRADE.TXT by the warning message below. I follow the setting in UPGRADE.TXT to set the call-limit for each user from value 1 - 10 but the state keeps changing as below. Anyone can tell me what kind of setting is necessary to reflect the state of the phone in queue? --no call-- 2200 has 0 calls (max unlimited) in 'rrmemory' strategy (3s holdtime), W:0, C:0, A:0, SL:0.0% within 120s Members: *CLI Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet No Callers --a call in queue-- 2200 has 1 calls (max unlimited) in 'rrmemory' strategy (3s holdtime), W:0, C:1, A:0, SL:100.0% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Unknown) has taken 1 calls (last was 7 secs ago) Callers: 1. SIP/2002-02c68200 (wait: 0:04, prio: 0) --call is answered-- [Jul 7 18:09:57] WARNING[30371]: app_queue.c:3023 try_calling: The device state of this queue member, Local/[EMAIL PROTECTED], is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. == Begin MixMonitor Recording SIP/2002-02c68200 ivrdevserver*CLI queue show 2200 has 0 calls (max unlimited) in 'rrmemory' strategy (10s holdtime), W:0, C:1, A:0, SL:100.0% within 120s Members: *CLI Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken 1 calls (last was 87 secs ago) No Callers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] background noise
Hi all, I would like to know how can I immunize the background noise in my case. Anyone can help? I have adjusted txgain rxgain in different value but the result is the same. ango Below is my configuration. asterisk1.4.21.1 zaptel1.4.11 addon1.4.7 TDM400 (FXOx4) There is a very large background noise when a call from sip to PSTN. Below is the test case. A - sip phone B - PSTN case 1: result is normal caller A1 callee A2 case 2: A hears B is ok but A hears much background noise from his/her area when A say nothing. The sound volume of A is very low when B hears A. caller A callee B case 3: same as case 2 caller B callee A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TCP UDP path not the same
HI, I got a one way audio when an ip phone dial to another ip phone in the same network. What I find is TCP UDP run different legs. Below is my configuration. asterisk (192.168.1.10) ipphone-A (192.168.1.111) ipphone-B (192.168.1.101) router (192.168.1.1) external IP (116.48.138.83) When A makes call to B, signal from A to router goes in the internal network. Then B pickup the call and I find that B will use external IP to reach the router. The signal from B finally can't reach to A. Below is a flow and you can see it involves using external IP. Is it related to the setting? Where and how to set it to make it work? U 192.168.1.10:5060 - 192.168.1.101:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0. Via: SIP/2.0/UDP 116.48.138.83:5060;branch=z9hG4bK612c1103;rport. From: 111 sip:[EMAIL PROTECTED];tag=as0a0b2a95. To: sip:[EMAIL PROTECTED]:5060. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: PBX. Max-Forwards: 70. Date: Fri, 13 Jun 2008 17:20:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 21200 21200 IN IP4 116.48.138.83. s=session. c=IN IP4 116.48.138.83. t=0 0. m=audio 19770 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. ___ -- 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] application sendtext
Hi, I want to send some text to the phone such that the phone can display the text on its display. I have tried to use SendText but it doesn't work. Does the phone need to support when asterisk issues the SendText application? ango ___ -- 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] queue autopause
Hi all, There is a setting called autopause in queue.conf to pause a queue member if they fail to answer a call. The autopause setting will pause the agent even when they are on the line. I want to know if it is possible to pause the queue member only when they don't answer after timeout? ango ___ -- 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] queue problem
I have a queue with the following setting. total queue member =30, autofill=1, timeout=25, monitor_format=wav49 asterisk 1.4.18 In busy hour, the loading of CPU reaches over 300%. At that moment, all members are occupied and many calls are waiting in the queue. There will be choppy and line cut at such high CPU loading. My questions: 1. What is the max capacity of a server to handle a queue in term of queue member and calls? 2. After every 25s, the call will be switched from agent to another agent. Can I do something, say execute a CLI or shell command before it switches to another agent? ango ___ -- 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] phone status question
Hi, A makes call to B. B has connection problem with the server (say, the lan cable is unplugged). 1: A --- server 2: A --- server 3: server B In 2, server will send the ring to A and it will hear ringing tone. In 3, server will try to connect B until timeout. My question is: A will still wait for B but B is physical unreachable. Can I set the number of retry time in server to try to reach the destination instead of the timeout? ___ -- 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] simple realtime question
HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango ___ -- 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] simple realtime question
It is what I found if realtime is applied. It also applies to all realtime setting, such as queue, voicemail etc. On Mon, May 5, 2008 at 8:03 PM, Benjamin Jacob [EMAIL PROTECTED] wrote: Last I was working on it, it did indeed NOT look at sip.conf with realtime architecture being used. But why take chances anyway? Move all the relevant conf files from /etc/asterisk to some other place to be safe. cheers - Ben. --- Rilawich Ango [EMAIL PROTECTED] wrote: HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango ___ -- 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 Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] segmentation fault
Segmentation fault occurs after executing the following cmd. Dial(SIP/[EMAIL PROTECTED]|35|Ttr) Is it a bug and how to fix it? Below is the core dump message converted by gdb. #0 0x068be1ad in realtime_peer (newpeername=0x1b37844 10.20.0.1, sin=0x0) at chan_sip.c:2547 #1 0x068becb3 in find_peer (peer=0x1b37844 10.20.0.1, sin=0x0, realtime=1) at chan_sip.c:2676 #2 0x068c0d4b in create_addr (dialog=0x9fb05b8, opeer=0x1b37e24 10.20.0.1) at chan_sip.c:2902 #3 0x069022d1 in sip_request_call (type=0x1b38010 SIP, format=256, data=0x1b38b08, cause=0x1b38c08) at chan_sip.c:15992 #4 0x0808c50a in ast_request (type=0x1b38010 SIP, format=256, data=0x1b38b08, cause=0x1b38c08) at channel.c:2994 #5 0x00df6d91 in dial_exec_full (chan=0x9f91f20, data=0x1b3af38, peerflags=0x1b38e04, continue_exec=0x0) at app_dial.c:1180 #6 0x00dfa3b6 in dial_exec (chan=0x9f91f20, data=0x1b3af38) at app_dial.c:1747 #7 0x080ce1ca in pbx_exec (c=0x9f91f20, app=0x9f37890, data=0x1b3af38) at pbx.c:537 #8 0x080d1f3b in pbx_extension_helper (c=0x9f91f20, con=0x0, context=0x9f92160 internal-admin, exten=0x9f921b0 104, priority=4, label=0x0, callerid=0x9f727b8 200, action=E_SPAWN) at pbx.c:1862 #9 0x080d3280 in ast_spawn_extension (c=0x9f91f20, context=0x9f92160 internal-admin, exten=0x9f921b0 104, priority=4, callerid=0x9f727b8 200) at pbx.c:2317 #10 0x080d37ac in __ast_pbx_run (c=0x9f91f20) at pbx.c:2419 #11 0x080d45c8 in pbx_thread (data=0x9f91f20) at pbx.c:2634 #12 0x08117094 in dummy_start (data=0x9f70fe8) at utils.c:865 #13 0x0076b45b in start_thread () from /lib/libpthread.so.0 #14 0x00bf124e in clone () from /lib/libc.so.6 ___ -- 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] one way audio after call transfer
Do you mean the problem is solved using asterisk 1.4.18? Are you using the setting as mine? Below is my setting. One way audio is a result after A B connected. PSTN (A)--1200P-- Asterisk -- GXP2000 --blind transfer -- Extension B You can see that involve many parties in the blind transfer operation. I am not sure the problem is related to 1200P, Asterisk or GXP2000. That's why I seeking the solution from any person who touch the same problem before. asterisk version: asterisk 1.4.15 zaptel 1.4.7 asterisk addons 1.4.5 On Thu, May 1, 2008 at 4:49 AM, Duncan Turnbull [EMAIL PROTECTED] wrote: I had a similar issue in 1.2 after transfer and we were using SIP only but an upgrade cured it We are now on 1.4.18 still without issues Cheers Duncan ___ -- 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] one way audio after call transfer
Hi all, Recently, I experienced one way audio after call transfer. incalling call (PSTN) A -- GXP2000 thro' zap --blind transfer-- destination B Finally A and B reach each others, but there is only one way audio. Anyone get the same experience before? How to solve the problem? Asterisk vesion: Asterisk 1.4.15 zaptel 1.4.7 asteriks-addon 1.4.5 ___ -- 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] database connections question
Hi all, In my understanding, we can use mssql as a database of asterisk thro' unixodbc. And we can easy using mysql (realtime) to do the same. Now, I want to keep 2 connections, one is mysql and one is mssql. Because both database have information that needed to be read from asterisk. Can I keep 2 connections in asterisk after it started, how? ango ___ -- 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] question about queue
Yup, I am using realtime queue. Do you mean the global setting in queue.conf is useless and you have to set every thing in each queue to activate the settings? If it is true, does it also apply to other realtime settings? On Tue, Apr 15, 2008 at 8:21 PM, Atis Lezdins [EMAIL PROTECTED] wrote: Hey, I just found out today that it doesn't work on Asterisk 1.4.19 (at least for realtime queues) if you have autofill=yes in queues.conf. However it works if you add it in queue settings for each queue, for realtime that would be ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED DEFAULT 1; For following this issue, see http://bugs.digium.com/view.php?id=12445 Regards, Atis On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ -- 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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] question about queue
Anyone can update me about the queue sticking by a caller? Is it solved in version 1.4.x? How? On Sat, Apr 12, 2008 at 9:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ -- 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] question about queue
Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ -- 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] question about queue
HI all, I have set up a queue with 2 members (A B). 1st call is waiting in the queue and a queue member A is ringing but don't take the call. Member A keeps ringing. Then 2nd call is also get into the queue but I found that queue member B doesn't ring. That's mean member B is available to take call but somehow it can't get a call in the queue. After that, member A takes the call of the 1st caller and member B gets ring. Question: -Why the 1st call will be stick the queue even there are many call behind? -Is it a bug of the queue or just a setting of the queue to solve the problem? 5000 has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:1, A:1, SL:100.0% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Unknown) has taken 1 calls (last was 76 secs ago) Callers: 1. SIP/2003-02cf0940 (wait: 0:47, prio: 0) 2. SIP/10.100.0.109-e4096dc0 (wait: 0:15, prio: 0) ___ -- 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] question about queue
Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes On Thu, Apr 10, 2008 at 8:57 PM, Don Pobanz [EMAIL PROTECTED] wrote: Rilawich Ango Thursday, April 10, 2008 3:28 AM I have set up a queue with 2 members (A B). 1st call is waiting in the queue and a queue member A is ringing but don't take the call. Member A keeps ringing. Then 2nd call is also get into the queue but I found that queue member B doesn't ring. That's mean member B is available to take call but somehow it can't get a call in the queue. After that, member A takes the call of the 1st caller and member B gets ring. What version of Asterisk are you using? This is a know issue/feature with version 1.2.x In version 1.4.x, set autofill=yes in queues.conf and calls will fill in as expected. Don Pobanz ___ -- 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] asterisk out of service
Below is the version I use in the system. asterisk: 1.4.18 addon 1.4.5 zaptel 1.4.8 It happened suddenly. My colleagues noticed all call were failed and asterisk process was hanged. There is no error in the system log (message) and no core dump at all. I just find the error message in the asterisk log (message) in the time of the accident occurred. On Thu, Mar 13, 2008 at 9:41 AM, Mike Fedyk [EMAIL PROTECTED] wrote: You'll need to post more info. Version and a scenario of what was happening at the time would be a good start... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Wednesday, March 12, 2008 6:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk out of service Hi all, I got the following message in the log yesterday. After that, no more in/out bound call can be made. What is the meaning of the message? ango [Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2367-d8062fb0! [Mar 12 09:26:17] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:33:15] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2327-dc32e4a0! [Mar 12 09:33:15] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:35:19] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/10.201.2.224-0e914380! [Mar 12 09:35:19] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] ___ -- 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] asterisk out of service
Hi all, I got the following message in the log yesterday. After that, no more in/out bound call can be made. What is the meaning of the message? ango [Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2367-d8062fb0! [Mar 12 09:26:17] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:33:15] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2327-dc32e4a0! [Mar 12 09:33:15] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Mar 12 09:35:19] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/10.201.2.224-0e914380! [Mar 12 09:35:19] ERROR[29565] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] ___ -- 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] from address modification
HI all, How can I modify the from address in sip message? Say, I will a sip account 1234. I want to change the from address in sip message of this sip account to 4321. From: 4321 sip:[EMAIL PROTECTED];tag=as5b42e6 ango ___ -- 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] restart asterisk daily
I have multiple queues in my case. Do you mean multiple queues is one of the reason to consume memory? How to only reset the queue stats? You will see asterisk behave its worst with multiple queues and heavy dialplan logic. I restart my boxes with queues everynight at midnight just to reset the queue stats displayed with show queue. ___ -- 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] restart asterisk daily
Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango ___ -- 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] restart asterisk daily
Actually, I donno it is a memory leak or not. I have a server only running asterisk. As time goes by, the free memory shown in the top is decreased. After I restart the asterisk, the free memory comes again. That's why I wonder if regular restart asterisk is necessary. Use a crontab to restart asterisk is a way to do it but you have to maintain a crontab. Is it possible to use logrotate instead? Or other better way? On Feb 13, 2008 3:26 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On 2/13/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango I have cron script that restarts daily by /etc/init.d/asterisk restart - that way asterisk is completely stopped and then started again. Long ago, for 1.2 this was helping to reduce crashes on working time. Anyway - i prefer to restart if i have some inactivity time anyway, rather than have a small chance that it will crash while lot's of calls are going trough. So, if you have timeframe when you know for sure that there won't be any calls, you can restart by initscript. Otherwise, if you have low call volume in night-time, you may modify safe_asterisk script to start asterisk anyway (even if it returns ok exit stats that means shutdown), and then stop asterisk by stop when convenient) Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] realtime warning
yes. On Feb 1, 2008 12:07 PM, Russell Bryant [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? Did you install res_config_mysql from asterisk-addons? -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] realtime warning
Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? ango ___ -- 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] volume problem
Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango ___ -- 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] pickup application failed
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup a call dialled from 1XX to 1YY (internal
network call). However, it is failed to pick up a call from PSTN
thro' TDM400 card. It seems I can't guess the correct context of it.
How can I know the context of the call using CLI? The default
context of the TDM400 is from-pstn but pickup still failed if I add
exten = _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
BLF_group_pickup.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
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Re: [asterisk-users] pickup application failed
Below is what I got from CLI
[Jan 7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 111.
On Jan 7, 2008 11:48 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup a call dialled from 1XX to 1YY (internal
network call). However, it is failed to pick up a call from PSTN
thro' TDM400 card. It seems I can't guess the correct context of it.
How can I know the context of the call using CLI? The default
context of the TDM400 is from-pstn but pickup still failed if I add
exten = _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
BLF_group_pickup.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
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[asterisk-users] auto dial and IVR
Hi, Is it possible to let asterisk auto dial out and play the IVR? How? i.e. -asterisk auto dial out (use outgoing folder?) -user pick the call -play IVR (1-for English, 2-for Chinese) -Then user can press the number to go through the level of IVR. ___ --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] dial, answered and then hangup
On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) Yes [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] dial, answered and then hangup
Phil, Thanks. The problem gone if it set to no. I want to know what is Polarity Reversal. I can't find it in the web. On Dec 18, 2007 12:02 AM, Phil Knighton [EMAIL PROTECTED] wrote: Rilawich, We use a TDM400P here in the UK, and if you set hanguponpolarityswitch=yes in zapata.conf, we get the same result. I think it is country specific, but try switching this to no and see what happens. This cured our problems. I have a note in zapata.conf (not sure if its from the release confs or from someone internal here) that says Fatal on TDM400P if set to yes. Good Luck Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: 17 December 2007 14:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial, answered and then hangup On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) Yes [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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 ___ --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] dial, answered and then hangup
Hi all,
I will a TDM card with FXO modules on it. Below is the dial plan.
When someone can 9123456, CLI will show dialing to 123456 and
answered. After answered, the call hangup. I would like to know what
will cause the case to happen. Anyone can give me some advice to
solve it?
exten = _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT})
exten = _9X.,n,Hangup
zapata.conf
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
;context=cs
channel = 1-8
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Re: [asterisk-users] dial, answered and then hangup
Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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] TDM400 hangup issue in China
Try to increase the value of busycount. It may help to solve the problem. On Dec 14, 2007 10:47 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; For me, I am in Kuwait and using the TDM22B and I used all the below settings and did not resolve my problem, I do not know if there is any other settings, or if there is a method to detect that no signaling is still existed on the, so we can do Hanup, the below settings used and did not resovle the hangup problem: busydetect=yes busycount=3 hanguponpolarityswitch=yes Any advise? Regards Bilal --- Thanks Steven and Giorgio that will work. There are a couple of other solutions depending on which telco you are connecting to and which exchange they are using. E1 offers more issues than PSTN. We are one of Digium's distributors in Asia with full technical support in China either out of our Hong Kong or Beijing offices. We have English and Chinese speaking people. Rupert Utteridge Director - Sales Marketing Digital Techniques (Asia) Limited Room 0209, Tower 2 Beijing Bright China Chang An Building 7 Jianguomen Nei Avenue Beijing 15 People's Republic of China Mobile: +86 136 8146 9331 Web: www.dtasia.net --- Thanks very much for the help Giorgio, I will give this a try today =) On Thu, 2007-12-13 at 11:40 +0100, gincantalupo wrote: Hi Steven, I do not live in China but I had the same problem. Try these 2 params inside zapata.conf: busydetect = yes hanguponpolarityswitch = yes It worked for me. Giorgio Incantalupo Steven O'Reilly wrote: Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help to hear about the configuration used. Best regards, Steven O'Reilly Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --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] TDM400 hangup issue in China
busydetect = yes hanguponpolarityswitch = yes Which of the two? busydetect will work almost always. But it is suboptimal: it may sotimes accidentally detect running calls. And it takes a few seconds to detect a hangup. Do you mean we need to adjust the value of busycount (larger than 5) to avoid suddenly hangup by the busydetect? If the provider sends you polarity events, you should be able to see them as debug messages of Asterisk (if debugging is turned on). Could you show us a sample for the message? We don't need to set hanguponpolarityswitch if there is no such event? ___ --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] merge gsm files
Hi, How can I merge 2 gsm files into a single file? I have tried to use soxmix as below but failed. soxmix 1.gsm 2.gsm 1-2.gsm ___ --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] line cut
Hi all, I have a TDM400 with all FXO in it. I can make outgoing out but the call will be dropped between 20-30mins suddenly. Below is the message shown in the log in the time the call drop. [Dec 10 23:23:32] DEBUG[3613] dsp.c: ast_dsp_busydetect detected busy, avgtone: 200, avgsilence 75 [Dec 10 23:23:32] DEBUG[3613] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/2-1 It said that it detect a busy tone and that hangup. I wonder why there is a busy tone during the call. Is it the setting problem? Anyone can give me some advices? ango ___ --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] Pickup cmd
HI,
I have tried to add the context but it still doesn't work.
On Dec 9, 2007 11:36 PM, F6HQZ [EMAIL PROTECTED] wrote:
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
your PickUp instruction or the reverse, or precise the context to use with
PickUp by adding it into the instruction line :
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**1XX,n,Hangup
Best Regards,
Francois BERGERET
France
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Re: [asterisk-users] Pickup cmd
thx. I got it.
On Dec 11, 2007 12:55 PM, dave cantera [EMAIL PROTECTED] wrote:
rilawich,
in the CLI type the following:
CLI dialplan show [EMAIL PROTECTED]
then
CLI dialplan show [EMAIL PROTECTED]
-or-
CLI dialplan show [EMAIL PROTECTED]
and see if * recognizes the x100 in either of those...
daveC
Rilawich Ango wrote:
HI,
I have tried to add the context but it still doesn't work.
On Dec 9, 2007 11:36 PM, F6HQZ [EMAIL PROTECTED] wrote:
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
your PickUp instruction or the reverse, or precise the context to use with
PickUp by adding it into the instruction line :
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**1XX,n,Hangup
Best Regards,
Francois BERGERET
France
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[asterisk-users] Pickup cmd
Hi all,
I have a GXP2000 with BLF configured. I follow the configuration
guide to enable the pickup cmd as follow and include it under
corresponding content.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:2})
exten = _**1XX,n,Hangup
The I press the single key to pickup the call to extension 100 when
there is a call to it. From CLI, I can see it issue **100 to asterisk
but failed to pickup the call.
-- Executing [EMAIL PROTECTED]:1] Pickup(SIP/102-08373480, 100)
in new stack
[Dec 7 16:47:42] NOTICE[31079]: app_directed_pickup.c:159
pickup_exec: No target channel found for 100.
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08373480, ) in new
stack
Anyone can tell me if I make something wrong for the pickup cmd?
asterisk version: 1.4.15
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[asterisk-users] dial in group
I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel = 1-8 ___ --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] dial in group
It works if it specified the port exactly plugged to PSTN. I want to clarify the dial command here. Dial(zap/g1/1234567) It will try channel 1, if it is busy, congested then it will try channel 2 and so on, right? I wonder if I don't plug the PSTN to channel 1, there should not be a dial tone on it. Why it still try channel 1 and make call using it? On Nov 25, 2007 5:00 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 24 Nov 2007, Rilawich Ango wrote: I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel = 1-8 You really only want channel = 3 here if it's only channel 3 that's plugged in. Gordon ___ --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] Dial problem
HI, I have 2 TDM400s plugged in a PC. I failed to use same channels to make a call to PSTN. It shows it can't establish connection after dial command issued. Below is the log. Actually, the call is established as I can hear voice from the called party but the softphone is still showing ringing. It seems the TDM card can't get an answered signal from PSTN. After 15 seconds, the call dropped because there is no answered signal. I want to know how to handle the problem? Is it related to settng? Can anyone tell me? [Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2001-0a0240c0, Zap/2/1872800|15) in new stack [Nov 23 01:23:11] DEBUG[5722] dsp.c: dsp busy pattern set to 0,0 [Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Dialing '1872800' [Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Deferring dialing... [Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Called 2/1872800 [Nov 23 01:23:14] DEBUG[5722] chan_zap.c: Done dialing, but waiting for progress detection before doing more... [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Nobody picked up in 15000 ms [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Hungup 'Zap/2-1' [Nov 23 01:23:27] NOTICE[5722] cdr.c: CDR on channel 'Zap/2-1' not posted [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/2001-0a0240c0, ) in new stack [Nov 23 01:23:27] VERBOSE[5722] logger.c: == Spawn extension (internal-admin, 91872800, 2) exited non-zero on 'SIP/2001-0a0240c0' [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/2001-0a0240c0, ) in new stack [Nov 23 01:23:27] VERBOSE[5722] logger.c: == Spawn extension (internal-admin, h, 1) exited non-zero on 'SIP/2001-0a0240c0' ___ --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] quality after call transfer
Hi, We are using attended call transfer to transfer the call. In the direct call, the quality of the voice and dtmf are acceptable. After transfer, the quality becomes worst. Voice can't be heard clearly and dtmf wrong detection will occur sometime. I wonder call transfer will affect he quality of the call. Anyone has same experience? Anything to do in asterisk level can get a better quality after call transfer? ___ --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] DTMF Problem
Does it help to turn on dtmf log in each servers? On Nov 16, 2007 5:01 PM, 木木 [EMAIL PROTECTED] wrote: I think you haven't capture the packet from the beginning of the call. You must capture the SIP packets. And the wireshark will recognise the packets as RTP. 木木 2007-11-16 发件人： Benjamin Jacob 发送时间： 2007-11-16 12:55:51 收件人： Asterisk Users Mailing List - Non-Commercial Discussion 抄送： 主题： Re: [asterisk-users] DTMF Problem for UDP tcpdump -nnXs 0 udp -i eth0 -w name.cap Btw, a pcap file (created on a linux server using tcpdump) capturing the RTP(udp) traffic opened up in wireshark, wireshark doesn't really format(or recognize) the packets as RTP, unlike the capture done live from a wireshark configured to capture RTP traffic. In the former, wireshark shows up everything as UDP and I have to do a lot of manual parsing to find out the type etc in the packets captured. Am I missing some config on wireshark here? TiA - Ben. ľľ wrote: You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 *发件人：* Doug *发送时间：* 2007-11-16 00:53:15 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No *抄送：* *主题：* Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] dtmf detection
Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA -- asterisk -- phoneB phoneA (music on hold), phoneB --attended call transfer-- phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause the wrong dtmf detection. ___ --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] function voicemailmain
Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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] function voicemailmain
You mean modify the source? Could you give me an example, say I wrong to remove advance option? On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: vi app_voicemail.c On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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] __sip_xmit problem
I got the cause of the problem. I set canreinvite=yes and the mentioned error gone. On Nov 10, 2007 12:27 AM, Steve Davies [EMAIL PROTECTED] wrote: On 11/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Steve Davies wrote: I would hazard that it is the port number of '0' that is causing the problem. It should not be necessary, but have you tried Dial(SIP/[EMAIL PROTECTED]:5060) To see if it helps. If so, it is probably symptomatic of a config error elsewhere. I seem to remember seeing reference to a similar problem on the list a couple of months back, but did not find it with a cursory search. You do not need to specify the port number. Asterisk will default to sending to UDP port 5060 if you don't specify a port. I know that, and you know that, and we all know that, but this is part of the wider diagnostic process to eliminate a bug :) I still suggest that it is tried to exclude it. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] crash
Hi all,
I have seen a lot of message talking about asterisk crashed when
using queue and mixmonitor together. I do use both in our system and
also get the crash (segfault) randomly. I don't know it is related to
the reason above as I have no idea about how it happened. I get the
core dump below. If anybody has any idea about the root cause of the
crash, please tell me.
Asterisk 1.4.13
Zaptel 1.4.5.1
Libpri 1.4.1
Addons 1.4.4
#0 0x0044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69
69 if (name[0] == '_') {
(gdb) bt
#0 0x0044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69
#1 0x0049948f in pbx_builtin_setvar_helper (chan=0x2aaac801a890,
name=0x2aaab69395a8 RTPAUDIOQOS,
value=0x2aaac80ecf20
ssrc=1967815032;themssrc=917073588;lp=61288;rxjitter=0.000165;rxcount=3668;txjitter=0.005142;txcount=1515;rlp=0;rtt=3.924000)
at pbx.c:5825
#2 0x2aaab6925a94 in handle_request_bye (p=0x2aaac80ba4e0, req=0x40255b10)
from /usr/lib/asterisk/modules/chan_sip.so
#3 0x2aaab69291ca in handle_request (p=0x2aaac80ba4e0, req=0x40255b10,
sin=0x40255b00, recount=0x40255af4, nounlock=0x40255af8)
from /usr/lib/asterisk/modules/chan_sip.so
#4 0x2aaab6929bb6 in sipsock_read (id=0x10b53330, fd=14, events=1,
ignore=0x0) from /usr/lib/asterisk/modules/chan_sip.so
#5 0x00474cd2 in ast_io_wait (ioc=0x10b50340, howlong=203) at io.c:279
#6 0x2aaab692a658 in do_monitor (data=0x0)
from /usr/lib/asterisk/modules/chan_sip.so
#7 0x004d4b8f in dummy_start (data=0x10b532c0) at utils.c:806
#8 0x003eaa6061b5 in start_thread () from /lib64/libpthread.so.0
#9 0x003ea9acd36d in clone () from /lib64/libc.so.6
#10 0x in ?? ()
(gdb)
___
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