Re: [asterisk-users] Playback while dialing out
A(x) does not accomplish this. It completes the playback and then dials. What I would like is that dialing should start in parallel and playback should stop as soon as early media or ringing starts. Similarly, music-on-hold is not an option, it's too hard coded, I like to be able to change playback file dynamically. Any hints?? On Fri, Aug 19, 2011 at 7:14 AM, Eric Wieling wrote: > Take a look at the A(x) and m options to dial. In the Asterisk CLI "core > show application dial" for a the docs to Dial(). > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin > Sent: Thursday, August 18, 2011 9:12 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Playback while dialing out > > Hi, please help me with dialplan below. > > My current dialplan looks like this, it plays a file and then connects the > caller to my phone by dialing out. As you can see, it waits for file to be > played completely before dialing out. What I would really like is that it > should play the file (preferably repetitively) and simultaneously dial out > the number, playback should stop as soon as dial answers or early media > detected. > > exten => 500,1,Answer > exten => 500,2,Playback(wait-while-we-connect-you) > exten => 500,3,Dial(SIP/14085551234@myprovider) > > How do I make it work? > > -- > _ > -- 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] Can't use SmartVoip/JustVoip after update
I was using Asterisk with SmartVoip and JustVoip for a long time running on CentOS 5. Recently I upgraded to CentOS 6 with a fresh build of Asterisk 1.8.5.0 Using the same firewall and SIP configuration as before now I can't dial out: retrans_pkt: Retransmission timeout reached on transmission ... Packet timed out after 6400ms with no response retrans_pkt: Hanging up call ... - no reply to our critical packet Port 5060 UDP incoming from NET to FW is enabled and not forwarded, all outgoing ports are enabled to the NET. I'm lost. Any idea, how to fix it? Thanks... Béla -- _ -- 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] Playback while dialing out
Hi, you can configure a new music on hold, example: nano /etc/asterisk/musiconhold.conf [default1] mode=files directory=moh1 and put the audio file in this directory; then change your dialplan like: exten => 500,1,NoOp exten => 500,2,Dial(SIP/14085551234@myprovider,m(default1)) exten => 503,3,Hangup Regards -- _ -- 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] How to get presence using AMI
any answer on below.. On 08/18/2011 03:50 PM, Nikhil wrote: Hi Using AMI how can I get the presence feature.Below are the requirement. --> List of all users in the PBX including analog and SIP including registration status. --> Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil -- _ -- 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] Possible Bug? .call files executing multiple times
Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call && mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, -- Brandon -- _ -- 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] Possible Bug? .call files executing multiple times
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call && mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a "self-logger"; the file won't "go away" until Asterisk "senses" that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Possible Bug? .call files executing multiple times
We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call&& mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a "self-logger"; the file won't "go away" until Asterisk "senses" that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Possible Bug? .call files executing multiple times
Danny, Now that I notice, MaxRetries has a default of 0 anyway, so not setting it should have not retried at all anyway. Although I did set it to 0 manually and still get the double calls. Brandon On 08/19/2011 09:31 AM, Brandon Phelps wrote: We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call&& mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a "self-logger"; the file won't "go away" until Asterisk "senses" that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Danny, Now that I notice, MaxRetries has a default of 0 anyway, so not setting it should have not retried at all anyway. Although I did set it to 0 manually and still get the double calls. Brandon On 08/19/2011 09:31 AM, Brandon Phelps wrote: We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call&& mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a "self-logger"; the file won't "go away" until Asterisk "senses" that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Outbound Dial
Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal -- _ -- 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] Outbound Dial
This is a simple call file task. First of all, I would convert the MP3 file to wav format. Then just use this call file and you're good to go (1 file for each PRI line) Line 1 extension: 170 Channel: DAHDI/R1/# app: Playback Application: Playback Data: /var/lib/asterisk/sounds/en/tt-monkeys Line 2 extension: 170 Channel: DAHDI/R2/# app: Playback Application: Playback Data: /var/lib/asterisk/sounds/en/tt-monkeys -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan Sent: Friday, August 19, 2011 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Outbound Dial Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal -- _ -- 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] Playback while dialing out
I am not sure you even read my mail, no music on hold option - it should work dynamically with any file. On Fri, Aug 19, 2011 at 6:18 PM, bakko wrote: > Hi, > > you can configure a new music on hold, example: > > nano /etc/asterisk/musiconhold.conf > > [default1] > mode=files > directory=moh1 > > and put the audio file in this directory; then change your dialplan like: > > exten => 500,1,NoOp > exten => 500,2,Dial(SIP/14085551234@myprovider,m(default1)) > exten => 503,3,Hangup > > Regards > > -- > _ > -- 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] Gtalk channel problem
We have been using gtalk channel from a long time now. It was working fine so far but from yesterday we are having problem. When gtalk destination is dialed and even answered, channel remains in Ringing state. Is there anything changed on google side? Jim -- _ -- 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] Gtalk channel problem
Yep, looks like Google changed something. Try this: https://issues.asterisk.org/jira/browse/ASTERISK-18301 Fixed it for me. On Fri, Aug 19, 2011 at 11:09 PM, Jim Boykin wrote: > We have been using gtalk channel from a long time now. It was working > fine so far but from yesterday we are having problem. When gtalk > destination is dialed and even answered, channel remains in Ringing > state. Is there anything changed on google side? > > Jim > -- _ -- 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
