Re: [asterisk-users] Asterisk Management API
On 11 March 2010 21:09, Matt Riddell li...@venturevoip.com wrote: On 9/03/10 9:13 PM, Peter Childs wrote: Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) When you make an originate request it works like this: 1. Call is made to the Channel parameter. 2. When the Channel answers it connects the other end to the application/context/extension. So, send the channel to the SIP device and then the other end won't start till the SIP device picks up. Yes I got that, and it seams to work quite well, It does mean that its more difficult to actually have a call going to a dead phone when it gets sent from the wrong channel in error. 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Again, looks like you have the order of the channels round the wrong way. If you originated to a SIP device and sent the other end to the application PlayDTMF, then it would be sent to the SIP device (if that's what you want). I figured that out. It means that if you want to control your calls when in you own menus, you can't do it by send DTMF but need to use the underlining application/dial-plan. which makes things more complex than they should be. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
On 9/03/10 9:13 PM, Peter Childs wrote: Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) When you make an originate request it works like this: 1. Call is made to the Channel parameter. 2. When the Channel answers it connects the other end to the application/context/extension. So, send the channel to the SIP device and then the other end won't start till the SIP device picks up. 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Again, looks like you have the order of the channels round the wrong way. If you originated to a SIP device and sent the other end to the application PlayDTMF, then it would be sent to the SIP device (if that's what you want). -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
On 9 March 2010 07:58, Peter Childs pchi...@bcs.org wrote: On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... /O -- _ -- 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] Asterisk Management API
Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol standard. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
At an Asterisk CLI use the command manager show commands. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 5, 2010, at 1:50 AM, Peter Childs wrote: Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol standard. Peter. -- _ -- 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] Asterisk Management API
Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there is more information available. Any pointers will be greatly appreciated. I hope to document my findings on the Wiki once I have definative information. Thanks Umar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Management API
The best way to figure out the manager protocols is through looking at the manager.c source code and trial and error. Some things just don't behave the way you think they should, some things are not fully documented and some actions do not work in certain cercumstances while others will. And when you figure something new out, please put it in the Wiki, I've added a lot to the Wiki in the manager API section and it took me quite a while to figure out some of it. Good luck, MATT--- -Original Message- From: Umar Sear [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 08, 2005 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Management API Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there is more information available. Any pointers will be greatly appreciated. I hope to document my findings on the Wiki once I have definative information. Thanks Umar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Management API
Umar Sear wrote: Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there is more information available. Any pointers will be greatly appreciated. I hope to document my findings on the Wiki once I have definative information. Thanks Umar Not sure what your looking for but you can just parse the output of the following commands show queues,show agents ie == Action: command Command: show queues == Response: Follows jrq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:137, A:0, SL:50.4% within 0s Members: Agent/3041 has taken 137 calls (last was 10 secs ago) No Callers mwq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:127, A:0, SL:44.9% within 0s Members: Agent/3042 has taken 127 calls (last was 68 secs ago) No Callers shq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3006 has taken no calls yet No Callers rgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3009 has taken no calls yet No Callers bfq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/1978 has taken no calls yet No Callers erq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3033 has taken no calls yet No Callers dwq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:39, A:0, SL:51.3% within 0s Members: Agent/3007 has taken 39 calls (last was 4234 secs ago) No Callers dhq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:87, A:0, SL:50.6% within 0s Members: Agent/3011 has taken 87 calls (last was 219 secs ago) No Callers mgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3025 has taken no calls yet No Callers joq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3028 has taken no calls yet No Callers lsq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:106, A:0, SL:41.5% within 0s Members: Agent/3017 has taken 106 calls (last was 12 secs ago) No Callers dmq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3010 has taken no calls yet No Callers sgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:57, A:0, SL:50.9% within 0s Members: Agent/3008 has taken 57 calls (last was 4797 secs ago) No Callers bcq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/1674 has taken no calls yet No Callers thq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/181 has taken no calls yet No Callers default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s No Members No Callers --END COMMAND-- == Action: command Command: show agents == Response: Follows 181 (Tom Hill) not logged in (musiconhold is 'none') 1674 (Bill Carron) not logged in (musiconhold is 'none') 3011 (Danny Harrington) logged in on Zap/4-1 is idle (musiconhold is 'none') 3028 (Justin Orstad) not logged in (musiconhold is 'none') 3025 (Mike Gaglio) not logged in (musiconhold is 'none') 3007 (Derrick Wilson) not logged in (musiconhold is 'none') 3008 (Steven Greenlaw) not logged in (musiconhold is 'none') 3033 (Eric Ryan) not logged in (musiconhold is 'none') 1978 (Bill Fornville) not logged in (musiconhold is 'none') 3006 (Saba Horton) not logged in (musiconhold is 'none') 3009 (Rob Giannina) not logged in (musiconhold is 'none') 3041 (John Rowley) logged in on Zap/16-1 talking to Zap/41-1 (musiconhold is 'none') 3042 (Michelle Wilson) logged in on Zap/15-1 is idle (musiconhold is 'none') 3017 (Laura Sood) logged in on Zap/2-1 is idle (musiconhold is 'rock1') 3010 (David McBrayer) not logged in (musiconhold is 'rock1') --END COMMAND-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
