Re: [asterisk-users] 3-way calling for IAX channels
We are developing an softphone based on IAX client version 1.2 (my current SIP softphone has many eoors), but it doesn´t have a specific function for Conferencing (3-way calling) or to place the other party on HOLD. I´m trying to do it through the PBX because our softphone´s lack of functions. I´ll be gratefull for further comments. Thanks again, Daniel On Tue, Jul 22, 2008 at 11:49 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling for IAX channels
Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling for IAX channels
2008/7/23 MFH [EMAIL PROTECTED]: Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. I believe that response is slightly misleading - Asterisk does not support conferencing without using meetme, but Zaptel/DAHDI will emulate the PSTN flash/recall facility which looks a bit like a conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI channel types, the endpoint must manage the equivalent of a PSTN flash/recall conference. Anything cross-channel or otherwise more complex does indeed require app_meetme. Given that the OP was referring to IAX, I believe they will need app_meetme. Of course I could be wrong :) Steve ___ -- 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] 3-way calling for IAX channels
On Wednesday 23 July 2008 12:17:26 Steve Davies wrote: 2008/7/23 MFH [EMAIL PROTECTED]: Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. I believe that response is slightly misleading - Asterisk does not support conferencing without using meetme, but Zaptel/DAHDI will emulate the PSTN flash/recall facility which looks a bit like a conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI channel types, the endpoint must manage the equivalent of a PSTN flash/recall conference. Anything cross-channel or otherwise more complex does indeed require app_meetme. Given that the OP was referring to IAX, I believe they will need app_meetme. The interesting thing is that Zaptel/DAHDI is using exactly the same conferencing/audio mixing engine as app_meetme. Or more correctly, app_meetme is using the Zaptel/DAHDI engine for audio mixing. -- Tilghman ___ -- 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] 3-way calling for IAX channels
How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 ___ -- 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] 3-way calling for IAX channels
Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling for IAX channels
Thanks for answering Noah, There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling for IAX channels
Hi Daniel - There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling
Do u mean meetme? It is total different from my case. In meetme, everybody need to know and dial the conference room number to get into the conference room. In my case, party A,B,C may not know the conference number. A only knows B numbers and B only knows C numbers. On 9/28/07, Pamela Weis [EMAIL PROTECTED] wrote: it is probably not what you are looking for. but simply use a conference room of asterisk for those 1 line phones. pamela ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
On Friday 28 September 2007 09:16:14 Rilawich Ango wrote: Do u mean meetme? It is total different from my case. In meetme, everybody need to know and dial the conference room number to get into the conference room. In my case, party A,B,C may not know the conference number. A only knows B numbers and B only knows C numbers. I'm planning to do something similar, and i have created a prototype code for this. So my prototype works: 1) A dials B 2) B presses some key to launch DYNAMIC_FEATURE (features.conf) 3) the feature fires a script that joins both channels to conf room. 4) B presses some key to exit from conf, and get to specified exit context. 5) DISA() there gives a dialtone, and launches dial to C 7) B presses first key again to join both calls to the same conference. 8) B can repeat again from 4 to add more calls to conference. Now reading all this gave me idea thaht it could be better to merge 3, 4 and 5 so that if nobody is in conference, you probably want to add some more people to conference - so just don't add B there, but give DISA straight away. Also this wouldn't allow neither A or C to add somebody to the same conference, as conference's name would match B's extension - otherwise it would be hard to determine wich conference to add. Regards, Atis On 9/28/07, Pamela Weis [EMAIL PROTECTED] wrote: it is probably not what you are looking for. but simply use a conference room of asterisk for those 1 line phones. pamela ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
From the web site said: 3-way Calling: Normally implemented by the phone. Can I do it in asterisk? How? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
What do you mean? I just want to know whether there is a way to do the following. 1. A --calls -- B 2. A on hold, B --calls -- C 3. A, B and C connected to talk On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote: How are you going to do it without a phone? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
How are you going to do it without a phone? PaulH On Thu, 2007-09-27 at 18:34 +0800, Rilawich Ango wrote: From the web site said: 3-way Calling: Normally implemented by the phone. Can I do it in asterisk? How? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
Rilawich Ango wrote: What do you mean? I just want to know whether there is a way to do the following. 1. A --calls -- B 2. A on hold, B --calls -- C 3. A, B and C connected to talk On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote: How are you going to do it without a phone? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 your phone would need a Join feature or you could do it externally with AMI but that would be clumsy. Most Sip phones have a 3way calling option right on them. Anthony ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
Your procedure as written below, is perfect and works fine. I have used Snom, Aastra and Polycom phones at various times to do exactly as you describe. PaulH On Fri, 2007-09-28 at 09:49 +0800, Rilawich Ango wrote: What do you mean? I just want to know whether there is a way to do the following. 1. A --calls -- B 2. A on hold, B --calls -- C 3. A, B and C connected to talk On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote: How are you going to do it without a phone? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
That's easy if phone supports 3 ways call. However, phones in my company only have 1 line without join function. Is it possible to implement 3 ways call using Asterisk without phone support in my case? On 9/28/07, Anthony Francis [EMAIL PROTECTED] wrote: Rilawich Ango wrote: What do you mean? I just want to know whether there is a way to do the following. 1. A --calls -- B 2. A on hold, B --calls -- C 3. A, B and C connected to talk On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote: How are you going to do it without a phone? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 your phone would need a Join feature or you could do it externally with AMI but that would be clumsy. Most Sip phones have a 3way calling option right on them. Anthony ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3-way calling
it is probably not what you are looking for. but simply use a conference room of asterisk for those 1 line phones. pamela Rilawich Ango wrote: That's easy if phone supports 3 ways call. However, phones in my company only have 1 line without join function. Is it possible to implement 3 ways call using Asterisk without phone support in my case? On 9/28/07, Anthony Francis [EMAIL PROTECTED] wrote: Rilawich Ango wrote: What do you mean? I just want to know whether there is a way to do the following. 1. A --calls -- B 2. A on hold, B --calls -- C 3. A, B and C connected to talk On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote: How are you going to do it without a phone? PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 your phone would need a Join feature or you could do it externally with AMI but that would be clumsy. Most Sip phones have a 3way calling option right on them. Anthony ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3 way Calling
Hello, I have recently installed the TrixBOX CE 2.2.4. How can I make calls and use the 3 way calling? can it be done with any IP phone or softphone? should I do any special configuration on TrixBox? Regards, Seysan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3 way calling independent of phone hw.
I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1. party 1 calls party 2 2. either party 1 or 2 hits * on keypad 3. asterisk prompts for party 3's telephone number 4. asterisk dials party 3. 5. party 3 answers and is immediately added to 3-way call 6. the inviter has the option of pushing # to terminate party 3 (should the call only reach party 3's voicemail). Either that or a ways to do DISA from within the meet-me functionality. I can't imagine I'm the only person with this sort of requirement. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3-way calling MGCP capture
Hello, I wonder if anyone can send me a trace of a 3-way calling trace done in MGCP ? I know you can do it in Asterisk, but you have to install a patch. I do not mind having a capture of a 3-way calling done not only in Asterisk, but in some other soft switch, like VocalData. Thank you in advance, Olga Mill [EMAIL PROTECTED] _ Olga Mill( [EMAIL PROTECTED] ) _ __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 way calling
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I have three way calling on my Bell lines, so b4 Asterisk, 3 way calls were established by establishing call 1, pressing the Flash key, dialing the other party, and finally pressing the Flash again to join everyone. The three way call only used one line. I'm able to do three way calling with Asterisk using the Flash key as well, but I just discovered that when I do it w/ Asterisk (press the Flash key, dial the other party, press Flash again) BOTH of my POTS lines (I only have 2) get used up. Any ideas on what's wrong w/ my configuration? Essentially, I need to be able to send the Flash signal to Bell in order to establish the three way call on a single FXO line, but I also need Asterisk to act on the Flash signal in order to transfer calls. Any ideas? Thanks, Hugh ___ 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] 3-Way Calling in Asterisk
You just described a conference call which is supported by most phones. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of aram Sent: Tuesday, April 12, 2005 6:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] 3-Way Calling in Asterisk Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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] 3-Way Calling in Asterisk
i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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] 3-Way Calling in Asterisk
Anybody doing it with Grandstream handytone ATA 286? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven Sent: Miércoles, 13 de Abril de 2005 08:29 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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] 3-Way Calling in Asterisk
As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, April 13, 2005 8:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 3-Way Calling in Asterisk Anybody doing it with Grandstream handytone ATA 286? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven Sent: Miércoles, 13 de Abril de 2005 08:29 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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 ___ 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] 3-Way Calling in Asterisk
I do it all the time.. Just like a standard phone Call someone, flash hook, get second dial tone, call another person, flash hook and all three are connected.. I didn't have to do anything, this works fine.. The one caveat to this is I cannot get it to work on my analog line (Don't know how to send the zaptel driver a flash hook event), so it only works if I use my VOIP provider.. - Andre -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Allen Niven Sent: Wednesday, April 13, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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] 3-Way Calling in Asterisk
Wiley Siler wrote: As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference This is BS. 3-Way calling is supported on both Cisco and SIPura ATAs, using FLASH just like any other analog 3-way call. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] 3-Way Calling in Asterisk
Thats what I figured, eventhough GS says the ata can do it, I dont see how because of the reason you just explained. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Miércoles, 13 de Abril de 2005 10:40 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 3-Way Calling in Asterisk As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, April 13, 2005 8:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 3-Way Calling in Asterisk Anybody doing it with Grandstream handytone ATA 286? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven Sent: Miércoles, 13 de Abril de 2005 08:29 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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 ___ 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] 3-Way Calling in Asterisk
Yep. I was mistaken. Already clarified with someone elses post. I was thinking conference not 3-way using flash. Thanks, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, April 13, 2005 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk Wiley Siler wrote: As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conferen ce This is BS. 3-Way calling is supported on both Cisco and SIPura ATAs, using FLASH just like any other analog 3-way call. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] 3-Way Calling in Asterisk
I have to agree with eric, 3 way calling, in my situation is as easy as putting the caller on hold, flash, and take the call off hold. Of course the appropriate settings have to be on in the zapata.conf___ 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] 3-Way Calling in Asterisk
Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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] 3 way calling feature
Ah Qiang wrote: Has anyone able to implement the 3 way calling feature? I have a 3 way calling plan with my telco. what I need to do is to call a number ,flashhook,call a 2nd number,flashhook , then the two party will be able to talk to each other. I am doing all this on a FXO card and all the calls are outgoing pstn calls. Any idea of how todo this? We talked about this on IRC. The above description does not even have close to enough information for anyone to help you. You didn't mention that you want Asterisk to initate the calls, not you. You didn't mention that you would be on your cell phone. You didn't diagram anything. In fact, your description of what you want to do guarntees what nobody will be able to help you. Spend some TIME writing up a detailed description what what you are trying to do, things people have suggested that did not work. Oh, and stop using the term 3-way call. A three-way call is done using FXS ports. You are using all FXO and so the term does not really apply here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 way calling feature
Hi, Dear allHas anyone able to implement the 3 way calling feature? I have a 3 way calling plan with my telco. what I need to do is to call a number,flashhook,call a 2nd number,flashhook , then the two party will be able to talk to each other. I am doing all this on a FXO card and all the calls areoutgoing pstn calls. Any idea of how todo this? regards Yahoo! Messenger- Log on with your mobile phone!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 Way Calling on Snom Phones and Asterisk
Has anyone been able to get 3way/Conference working with the snom200 and Asterisk. According to the documentation for the phones the option should come up when you have two lines active on the snom phone. Unfortunately, I don't see this option appear and I am now beginning to wonder if this is a limitation of Asterisk. Does anyone have any suggestions? Any help would be greatly appreciated. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling
I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 3-way calling
The voip wiki says that three way calling is implemented on the client side. I searched google and found some previous threads on the Asterisk list that may be helpful to you: http://www.voip-info.org/wiki-Asterisk+PBX+functions http://lists.digium.com/pipermail/asterisk-users/2003-April/010840.html http://lists.digium.com/pipermail/asterisk-users/2004-July/055328.html http://lists.digium.com/pipermail/asterisk-users/2003-November/025556.html http://lists.digium.com/pipermail/asterisk-users/2003-December/030380.html Good luck - Brian I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 3-way calling
On Tue, 2004-09-14 at 17:04, Bill Hamlin wrote: I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? That works exactly as expected on Zap interfaces. For VoIP devices it's TOTALLY handled by the phone. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 3-way calling
That works exactly as expected on Zap interfaces. For VoIP devices it's TOTALLY handled by the phone. If you're extremely lucky :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling woes... Nasty static and inconsistent flash detection?
This is my setup: SPA-2000 - Asterisk - X101P (x4) - PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on hold and gives me a dial tone. Now if I call a second number that goes over the PSTN, and try to connect all 3 parties (hitting flash again) it works, but I get this terrible static on the line that blasts everyone in the ear making it impossible to talk. As soon as one of the remote parties hangs up, the static stops and I can continue to talk to that one person just fine. Here is a log of the output of the inconsistent flash detection, with my commentary: -- Executing Macro(SIP/705-49a2, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-49a2, local) in new stack -- Executing Dial(SIP/705-49a2, Zap/g1/7680989||t) in new stack -- Called g1/7680989 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/705-49a2 *** Testing the flash button a few times, making sure it puts the call on hold, which it did. -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' *** Intentionally ended the call, to immediately call back and try again. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-49a2' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- 49a2' -- Executing Macro(SIP/705-37c1, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-37c1, local) in new stack -- Executing Dial(SIP/705-37c1, Zap/g1/7680989||t) in new stack -- Called g1/7680989 Jun 25 16:56:12 WARNING[1191312304]: app_dial.c:338 wait_for_answer: Unable to forward frame -- Hungup 'Zap/1-1' *** Pressed flash, it hung up. So called back to try it again. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-37c1' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- 37c1' -- Executing Macro(SIP/705-bb8f, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-bb8f, local) in new stack -- Executing Dial(SIP/705-bb8f, Zap/g1/7680989||t) in new stack -- Called g1/7680989 -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' *** Pressed flash, it hung up. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-bb8f' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- bb8f' I don't have any logs of the 3-way in progress when I was getting the static. The static only happens when the two outgoing channels are ZAP. I wonder if the static has anything to do with the echo cancellation just committed in to CVS that I'm using? I'll have to test this theory when I get a chance tomorrow. I'm running CVS Head from yesterday. As well the SPA-2000's are running firmware 2.0.8, and there are no IRQ sharing conflicts. So, any ideas why Asterisk is sometimes putting the call on hold when flash is hit, and other times just hanging up? The flash button obviously works. As well, what would cause the nasty static when a 3-way call with 2 of the parties connected over the PSTN finally does work? Notable lines in my zapata.conf file: context=mainmenu signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes musiconhold=default faxdetect=both echocancel=yes echocancelwhenbridged=yes echotraining=800 -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 3-way calling bug
Sorry for the lack of info...here goes - We are using CVS from earlier this week. Our phones are fxs signalling and it is connected to asterisk via a channelbank and a TE410P card. I have put a bug into the bugtracker, it's ID is 687 thanks, Derek On Thu, 2003-12-18 at 19:19, John Todd wrote: Hi, I discovered a problem in asterisk with the following scenerio: 1) I make an outbound call 2) Called person answers phone 3) I hit the flashhook to initiate a 3-way call 4) I hear dial tone and called person is on hold 5) I hang up my phone 6) called person hangs up their phone 7) my phone starts ringing 8) I answer and no one is there, I hang up 9) endless loop between step 7 8 happens after this happens and this endless ringing loop begins asterisk cannot be stopped from within the console but must be killed with kill -9. Any help or insight into the matter would be greatly appreciated. Thanks, Derek What kind of phones? SIP? IAX? fxs? What version of CVS? What equipment? Please document a bit more, and perhaps this might need a report in the bugtracker (http://bugs.digium.com/) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling bug
Hi, I discovered a problem in asterisk with the following scenerio: 1) I make an outbound call 2) Called person answers phone 3) I hit the flashhook to initiate a 3-way call 4) I hear dial tone and called person is on hold 5) I hang up my phone 6) called person hangs up their phone 7) my phone starts ringing 8) I answer and no one is there, I hang up 9) endless loop between step 7 8 happens after this happens and this endless ringing loop begins asterisk cannot be stopped from within the console but must be killed with kill -9. Any help or insight into the matter would be greatly appreciated. Thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3-way calling bug
Hi, I discovered a problem in asterisk with the following scenerio: 1) I make an outbound call 2) Called person answers phone 3) I hit the flashhook to initiate a 3-way call 4) I hear dial tone and called person is on hold 5) I hang up my phone 6) called person hangs up their phone 7) my phone starts ringing 8) I answer and no one is there, I hang up 9) endless loop between step 7 8 happens after this happens and this endless ringing loop begins asterisk cannot be stopped from within the console but must be killed with kill -9. Any help or insight into the matter would be greatly appreciated. Thanks, Derek What kind of phones? SIP? IAX? fxs? What version of CVS? What equipment? Please document a bit more, and perhaps this might need a report in the bugtracker (http://bugs.digium.com/) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
