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From: Dov B. <dov...@gm...> - 2007-04-19 22:15:39
|
Hi, Is it possible to add a SIPHeader (application AddSipHeader on Asterisk) when I make a call through asterisk-java, for example, using the OriginateAction? I want to make an application that Dials to a VoiceXML application, which would read parameters from the SIP header... (Or doesn't anyone have a better idea for sending parameters through a Dial...) Thank you! Dov |
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From: Stefan R. <ste...@re...> - 2007-04-18 19:55:37
|
Rajpal Dangi wrote:
> Iam very new to Astersik community and exploring some ways to bridg=
e
> two chennels using either Manager API or FastAGI. Is it possible to do
> so? I don't see any action/command which I can call to do so.
With Asterisk 1.2 and 1.4 this is not possible. Asterisk 1.6 (or current
trunk) adds support for a Bridge action:
Action: Bridge
Synopsis: Bridge two channels already in the PBX
Privilege: command,all
Description: Bridge together two channels already in the PBX
Variables: ( Headers marked with * are required )
*Channel1: Channel to Bridge to Channel2
*Channel2: Channel to Bridge to Channel1
Tone: (Yes|No) Play courtesy tone to Channel 2
So for now the only option to go is to redirect both channels to a
MeetMe conference.
=3DStefan
--=20
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail: ste...@re...
Jabber: ste...@re...
Steuernummern 215/5140/1791 USt-IdNr. DE220701760
|
|
From: Rajpal D. <sip...@gm...> - 2007-04-18 17:21:17
|
Hi All,
Iam very new to Astersik community and exploring some ways to bridge two
chennels using either Manager API or FastAGI. Is it possible to do so? I
don't see any action/command which I can call to do so.
Thanks a lot,
/dan
|
|
From: Stefan R. <ste...@re...> - 2007-04-18 00:57:30
|
robert home wrote: > does any one know what happened to www.asterisk-java.org > or when it'll be back We had problems with the IN NS records at PSI. The problem is fixed now though it might still take a few hours for the changes to propagate. I am sorry for any inconvinience this outage may have caused and have provided a bonus article on Local/ channels to say sorry. The article include a nice diagram on how using Local channels and Originate relates to the events you see on the Manager API. =3DStefan P.S. If you still encounter problems please contact me off-list and I'll have a look if I still missed anything. --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
|
From: Vivek G. <vi...@in...> - 2007-04-16 06:32:46
|
Hi All, We are using asterisk-java to give a one ring call to a phone no. Simple flow is - 1. Originate call to a number using 1234567 as caller id 2. Catch the event of new callerid to get the channel id for phone no -1234567 3. Catch the event for new channel event with status ringing. - send the Hangup action 4. Exit But TimeoutException is occuring when we try to hangup. How to get rid of this exception? Please see attached the code and exception stack trace. Thanks in advance.. Regards Vivek |
|
From: robert h. <rls...@op...> - 2007-04-11 22:41:49
|
does any one know what happened to www.asterisk-java.org or when it'll = be back thanks |
|
From: Michael Y. <mic...@i9...> - 2007-04-04 20:51:27
|
I forgot to mention that this does not happen every time only about 30-40% of the time. Also I believe that the improper id is grabbed from a previous channel that has already been hung up but has the same channel name as the one placed in the meet me room. Michael Yara wrote: > I am having a problem with one of the meet me join events that I am > getting from asterisk java live when i redirect a channel and its linked > channel to a meet me conference room. > > These are the events that I am getting from asterisk java live: > > New Channel:--------------- > Name: AsyncGoto/SIP/4222-0978c1c0 > ID: 1175712744.1870 > > (AsyncGoto/SIP/4222-0978c1c0 gets renamed to SIP/4222-0978c1c0 and the > original SIP/4222-0978c1c0 gets renamed to a zombie and hangs up) > > New MeetMeUser:--------------- > Room: 42248109 > Channel: SIP/4222-0978c1c0 > ChannelId: 1175712732.1865 > UserNumber: 2 > RoomNumber: 42248109 > > > And these are the events that I am getting if I connect directly to the > manager: > > Event: Newchannel > Privilege: call,all > Channel: AsyncGoto/SIP/4222-0978c1c0 > State: Up > CallerID: <unknown> > CallerIDName: <unknown> > Uniqueid: 1175712744.1870 > > Event: MeetmeJoin > Privilege: call,all > Channel: SIP/4222-0978c1c0 > Uniqueid: 1175712744.1870 > Meetme: 42248109 > Usernum: 2 > > notice that the channel id of the asterisk-java MeetMeUser event is not > the same as the id that is given by the manager > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
|
From: Michael Y. <mic...@i9...> - 2007-04-04 19:48:41
|
I am having a problem with one of the meet me join events that I am getting from asterisk java live when i redirect a channel and its linked channel to a meet me conference room. These are the events that I am getting from asterisk java live: New Channel:--------------- Name: AsyncGoto/SIP/4222-0978c1c0 ID: 1175712744.1870 (AsyncGoto/SIP/4222-0978c1c0 gets renamed to SIP/4222-0978c1c0 and the original SIP/4222-0978c1c0 gets renamed to a zombie and hangs up) New MeetMeUser:--------------- Room: 42248109 Channel: SIP/4222-0978c1c0 ChannelId: 1175712732.1865 UserNumber: 2 RoomNumber: 42248109 And these are the events that I am getting if I connect directly to the manager: Event: Newchannel Privilege: call,all Channel: AsyncGoto/SIP/4222-0978c1c0 State: Up CallerID: <unknown> CallerIDName: <unknown> Uniqueid: 1175712744.1870 Event: MeetmeJoin Privilege: call,all Channel: SIP/4222-0978c1c0 Uniqueid: 1175712744.1870 Meetme: 42248109 Usernum: 2 notice that the channel id of the asterisk-java MeetMeUser event is not the same as the id that is given by the manager |
|
From: Markus F. <mar...@ts...> - 2007-04-03 09:12:27
|
Thanks, i will try it next time. Markus=20 -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Dienstag, 3. April 2007 11:08 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API That's where the dialplan goes upon hangup in the current context. See http://www.voip-info.org/wiki/index.php?page=3DAsterisk+h+extension or http://www.voip-info.org/tiki-index.php?page=3DAsterisk%20config%20extens= ions.conf under the "Predefined extension names" section. Maybe you could redirect the other party's leg there if it has not yet been hung up (once again, not sure it'd work) ? Gaetan Markus Floegel wrote: > What do you meen with the "h" extension? > =20 > > ________________________________ > > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet > Gesendet: Dienstag, 3. April 2007 10:06 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > > Ok I see. In our application the operator links (transfer/redirect) = the channels in the application as she has no phone to hang up, and the = pbx hangs up the operator phone after the redirect occurs so we don't = have this problem; > In your case you need a transfer supervised by the pbx, not by your = application. Indeed you'll need somethink like a meetme to keep = everybody connected.=20 > > Have you tried to do the bridge in the "h" extension in the operator = context so you redirect the other party before its channel gets closed, = using channel variables (not sure it could work) ? > > Gaetan > > Markus Floegel wrote:=20 > > Hi Gaetan, > =09 > Thanks for the answer! > =09 > =20 > > the operator line is > hung up by the system even before she puts the phone down (in fact = the > phone is an headset with autoanswer in our case). > =20 > > =09 > In my application, the operator manualy hangs up the phone (using = hardware) and then the other two channels should be connected. > But there is the problem. The the third party is disconnected too, = when the operator hangs up the phone. > =09 > I tried to catch the HangUpEvent or UnLinkEvent of the third channel = and then redirect the third channel to the parking channel but it = doesn=B4t work... > =09 > Markus > =09 > =20 > =09 > -----Urspr=FCngliche Nachricht----- > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet > Gesendet: Dienstag, 3. April 2007 09:19 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager = API > =09 > Hi Markus > =09 > If the channel is hung up when the operator hangs up the phone = chances > are that the parking didn't work at all, because in our application = as > soon as we redirect the other party to the parking, the operator line = is > hung up by the system even before she puts the phone down (in fact = the > phone is an headset with autoanswer in our case). > =09 > In the dialplan I simply put this kind of thing: > =09 > [CUSTOMPARK] > exten =3D> *101,1,StopMusicOnHold() > exten =3D> *101,2,StartMusicOnHold() > exten =3D> *101,3,ParkSilent() > =09 > Then I redirect the third party's leg to the extension > (CUSTOMPARK,*101,1) using AMI. > =09 > Please note we don't use the parking extension declared in > features.general to avoid any interference with the stock parking > application, and that as a consequence the *101 is not automatically > inserted in the parkingext context by res_features. We use (and = redirect > to) our own separate context. > =09 > The only reason I see is that your parking extensions was not = declared > in your context so the redirection simply fails as the dialplan can't > reach the extension. If you where using the stock parking application > and configuration, maybe you forgot to include the parkingext > (sub-)context in you incoming(zap ?) context ? > =09 > =09 > N.B: The stop/start MOH thing is here because with at least asterisk > 1.2.9 we had problem with the channel crashing when coming from a = queue > where MOH was already playing, unless we used mpg123. That had = something > to do with codec translation changes on the fly iirc; > =09 > Gaetan > =09 > Markus Floegel wrote: > =20 > > Hi, > =09 > I tried something like this before, but i had the problem that the = third party was disconnected when the operator hangs up the phone. > So i can=B4t redirect the channel of the third party to the parking = channel. I alway gets an HangupEvent and the phone of the third party = was disconnected. > How do you implement this? > =09 > thanks > =09 > =09 > =20 > =09 > ________________________________ > =09 > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet > Gesendet: Montag, 2. April 2007 11:02 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager = API > =09 > =09 > Hi, > =09 > We managed to implement this using (well, abusing) the parking = application. That way we have no meetme running for each attended call, = the channels are always simply bridged. > =09 > - We redirect the leg of the caller to the parking extension. The = operator leg is then dropped and the caller has MOH. In return we get = the parking position of the call. > - We then call the other party, announce the call. > - If he accepts the call, we redirect his current leg to the parking = lot extension the call is parked at (and the operator leg is dropped = once again). The other party gets the right tone just before the = channels are bridged by the parking application. > - If he refuses the call we redirect our own leg to the parking = extension so we get the caller back. > =09 > The only problem is that with the stock Park application, the caller = is announced it's own parking lot extension so we made a tiny ast = application, called "ParkSilent". That's just a copy&paste of the stock = parking main function with second party channel set to null instead of = the caller channel. This can be easily compiled against at least 1.2.9 - = 1.2.16 without recompiling the whole asterisk (in fact it's even binary = compatible as it links dynamically against the stock parking). > =09 > Regards > =09 > Gaetan > =09 > =09 > =09 > =09 > Markus Floegel wrote:=20 > =09 > I think such a Action in the Manager will be very good. > The workaround with the meetMe Application is not so fine, i think. > =09 > Markus > =09 > =09 > =09 > -----Urspr=FCngliche Nachricht----- > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Stefan Reuter > Gesendet: Freitag, 30. M=E4rz 2007 10:25 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager = API > =09 > Markus Floegel wrote: > =20 > =09 > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from = Asterisk? > =20 > =09 > =09 > I've seen this question quite a few times now without any suitable = answer. > Maybe its time to implement a Manager Action for this feature and = submit > it as a patch to Asterisk? > Anybody want's to take this? > =09 > I would be happy to support that new action in Asterisk-Java. > =09 > =3DStefan > =09 > =20 > =09 > =09 > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > =09 > =20 > =20 > > =09 > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > =09 > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > =09 > =20 > > > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > =20 -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share = your opinions on IT & business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
|
From: Gaetan M. <gm...@ea...> - 2007-04-03 09:07:55
|
That's where the dialplan goes upon hangup in the current context. See http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension or http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf under the "Predefined extension names" section. Maybe you could redirect the other party's leg there if it has not yet been hung up (once again, not sure it'd work) ? Gaetan Markus Floegel wrote: > What do you meen with the "h" extension? > > > ________________________________ > > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet > Gesendet: Dienstag, 3. April 2007 10:06 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > > Ok I see. In our application the operator links (transfer/redirect) the channels in the application as she has no phone to hang up, and the pbx hangs up the operator phone after the redirect occurs so we don't have this problem; > In your case you need a transfer supervised by the pbx, not by your application. Indeed you'll need somethink like a meetme to keep everybody connected. > > Have you tried to do the bridge in the "h" extension in the operator context so you redirect the other party before its channel gets closed, using channel variables (not sure it could work) ? > > Gaetan > > Markus Floegel wrote: > > Hi Gaetan, > > Thanks for the answer! > > > > the operator line is > hung up by the system even before she puts the phone down (in fact the > phone is an headset with autoanswer in our case). > > > > In my application, the operator manualy hangs up the phone (using hardware) and then the other two channels should be connected. > But there is the problem. The the third party is disconnected too, when the operator hangs up the phone. > > I tried to catch the HangUpEvent or UnLinkEvent of the third channel and then redirect the third channel to the parking channel but it doesn´t work... > > Markus > > > > -----Ursprüngliche Nachricht----- > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet > Gesendet: Dienstag, 3. April 2007 09:19 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > Hi Markus > > If the channel is hung up when the operator hangs up the phone chances > are that the parking didn't work at all, because in our application as > soon as we redirect the other party to the parking, the operator line is > hung up by the system even before she puts the phone down (in fact the > phone is an headset with autoanswer in our case). > > In the dialplan I simply put this kind of thing: > > [CUSTOMPARK] > exten => *101,1,StopMusicOnHold() > exten => *101,2,StartMusicOnHold() > exten => *101,3,ParkSilent() > > Then I redirect the third party's leg to the extension > (CUSTOMPARK,*101,1) using AMI. > > Please note we don't use the parking extension declared in > features.general to avoid any interference with the stock parking > application, and that as a consequence the *101 is not automatically > inserted in the parkingext context by res_features. We use (and redirect > to) our own separate context. > > The only reason I see is that your parking extensions was not declared > in your context so the redirection simply fails as the dialplan can't > reach the extension. If you where using the stock parking application > and configuration, maybe you forgot to include the parkingext > (sub-)context in you incoming(zap ?) context ? > > > N.B: The stop/start MOH thing is here because with at least asterisk > 1.2.9 we had problem with the channel crashing when coming from a queue > where MOH was already playing, unless we used mpg123. That had something > to do with codec translation changes on the fly iirc; > > Gaetan > > Markus Floegel wrote: > > > Hi, > > I tried something like this before, but i had the problem that the third party was disconnected when the operator hangs up the phone. > So i can´t redirect the channel of the third party to the parking channel. I alway gets an HangupEvent and the phone of the third party was disconnected. > How do you implement this? > > thanks > > > > > ________________________________ > > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet > Gesendet: Montag, 2. April 2007 11:02 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > > Hi, > > We managed to implement this using (well, abusing) the parking application. That way we have no meetme running for each attended call, the channels are always simply bridged. > > - We redirect the leg of the caller to the parking extension. The operator leg is then dropped and the caller has MOH. In return we get the parking position of the call. > - We then call the other party, announce the call. > - If he accepts the call, we redirect his current leg to the parking lot extension the call is parked at (and the operator leg is dropped once again). The other party gets the right tone just before the channels are bridged by the parking application. > - If he refuses the call we redirect our own leg to the parking extension so we get the caller back. > > The only problem is that with the stock Park application, the caller is announced it's own parking lot extension so we made a tiny ast application, called "ParkSilent". That's just a copy&paste of the stock parking main function with second party channel set to null instead of the caller channel. This can be easily compiled against at least 1.2.9 - 1.2.16 without recompiling the whole asterisk (in fact it's even binary compatible as it links dynamically against the stock parking). > > Regards > > Gaetan > > > > > Markus Floegel wrote: > > I think such a Action in the Manager will be very good. > The workaround with the meetMe Application is not so fine, i think. > > Markus > > > > -----Ursprüngliche Nachricht----- > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Stefan Reuter > Gesendet: Freitag, 30. März 2007 10:25 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > Markus Floegel wrote: > > > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from Asterisk? > > > > I've seen this question quite a few times now without any suitable answer. > Maybe its time to implement a Manager Action for this feature and submit > it as a patch to Asterisk? > Anybody want's to take this? > > I would be happy to support that new action in Asterisk-Java. > > =Stefan > > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > |
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From: Markus F. <mar...@ts...> - 2007-04-03 08:27:21
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What do you meen with the "h" extension? =20 ________________________________ Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Dienstag, 3. April 2007 10:06 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API Ok I see. In our application the operator links (transfer/redirect) the = channels in the application as she has no phone to hang up, and the pbx = hangs up the operator phone after the redirect occurs so we don't have = this problem; In your case you need a transfer supervised by the pbx, not by your = application. Indeed you'll need somethink like a meetme to keep = everybody connected.=20 Have you tried to do the bridge in the "h" extension in the operator = context so you redirect the other party before its channel gets closed, = using channel variables (not sure it could work) ? Gaetan Markus Floegel wrote:=20 Hi Gaetan, =09 Thanks for the answer! =09 =20 the operator line is hung up by the system even before she puts the phone down (in fact = the phone is an headset with autoanswer in our case). =20 =09 In my application, the operator manualy hangs up the phone (using = hardware) and then the other two channels should be connected. But there is the problem. The the third party is disconnected too, when = the operator hangs up the phone. =09 I tried to catch the HangUpEvent or UnLinkEvent of the third channel = and then redirect the third channel to the parking channel but it = doesn=B4t work... =09 Markus =09 =20 =09 -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Dienstag, 3. April 2007 09:19 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API =09 Hi Markus =09 If the channel is hung up when the operator hangs up the phone chances are that the parking didn't work at all, because in our application as soon as we redirect the other party to the parking, the operator line = is hung up by the system even before she puts the phone down (in fact the phone is an headset with autoanswer in our case). =09 In the dialplan I simply put this kind of thing: =09 [CUSTOMPARK] exten =3D> *101,1,StopMusicOnHold() exten =3D> *101,2,StartMusicOnHold() exten =3D> *101,3,ParkSilent() =09 Then I redirect the third party's leg to the extension (CUSTOMPARK,*101,1) using AMI. =09 Please note we don't use the parking extension declared in features.general to avoid any interference with the stock parking application, and that as a consequence the *101 is not automatically inserted in the parkingext context by res_features. We use (and = redirect to) our own separate context. =09 The only reason I see is that your parking extensions was not declared in your context so the redirection simply fails as the dialplan can't reach the extension. If you where using the stock parking application and configuration, maybe you forgot to include the parkingext (sub-)context in you incoming(zap ?) context ? =09 =09 N.B: The stop/start MOH thing is here because with at least asterisk 1.2.9 we had problem with the channel crashing when coming from a queue where MOH was already playing, unless we used mpg123. That had = something to do with codec translation changes on the fly iirc; =09 Gaetan =09 Markus Floegel wrote: =20 Hi, =09 I tried something like this before, but i had the problem that the = third party was disconnected when the operator hangs up the phone. So i can=B4t redirect the channel of the third party to the parking = channel. I alway gets an HangupEvent and the phone of the third party = was disconnected. How do you implement this? =09 thanks =09 =09 =20 =09 ________________________________ =09 Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Montag, 2. April 2007 11:02 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API =09 =09 Hi, =09 We managed to implement this using (well, abusing) the parking = application. That way we have no meetme running for each attended call, = the channels are always simply bridged. =09 - We redirect the leg of the caller to the parking extension. The = operator leg is then dropped and the caller has MOH. In return we get = the parking position of the call. - We then call the other party, announce the call. - If he accepts the call, we redirect his current leg to the parking = lot extension the call is parked at (and the operator leg is dropped = once again). The other party gets the right tone just before the = channels are bridged by the parking application. - If he refuses the call we redirect our own leg to the parking = extension so we get the caller back. =09 The only problem is that with the stock Park application, the caller = is announced it's own parking lot extension so we made a tiny ast = application, called "ParkSilent". That's just a copy&paste of the stock = parking main function with second party channel set to null instead of = the caller channel. This can be easily compiled against at least 1.2.9 - = 1.2.16 without recompiling the whole asterisk (in fact it's even binary = compatible as it links dynamically against the stock parking). =09 Regards =09 Gaetan =09 =09 =09 =09 Markus Floegel wrote:=20 =09 I think such a Action in the Manager will be very good. The workaround with the meetMe Application is not so fine, i think. =09 Markus =09 =09 =09 -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Stefan Reuter Gesendet: Freitag, 30. M=E4rz 2007 10:25 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager = API =09 Markus Floegel wrote: =20 =09 I implemented the same now im my project. But is there no way to use the internal 'attended transfer' from = Asterisk? =20 =09 =09 I've seen this question quite a few times now without any suitable = answer. Maybe its time to implement a Manager Action for this feature and = submit it as a patch to Asterisk? Anybody want's to take this? =09 I would be happy to support that new action in Asterisk-Java. =09 =3DStefan =09 =20 =09 =09 = -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to = share your opinions on IT & business topics through brief surveys-and earn cash = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users =09 =20 =20 =09 = -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share = your opinions on IT & business topics through brief surveys-and earn cash = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users =09 = -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share = your opinions on IT & business topics through brief surveys-and earn cash = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users =09 =20 |
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From: Gaetan M. <gm...@ea...> - 2007-04-03 08:05:53
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Ok I see. In our application the operator links (transfer/redirect) the channels in the application as she has no phone to hang up, and the pbx hangs up the operator phone after the redirect occurs so we don't have this problem; In your case you need a transfer supervised by the pbx, not by your application. Indeed you'll need somethink like a meetme to keep everybody connected. Have you tried to do the bridge in the "h" extension in the operator context so you redirect the other party before its channel gets closed, using channel variables (not sure it could work) ? Gaetan Markus Floegel wrote: > Hi Gaetan, > > Thanks for the answer! > > >>> the operator line is >>> hung up by the system even before she puts the phone down (in fact the >>> phone is an headset with autoanswer in our case). >>> > > In my application, the operator manualy hangs up the phone (using hardware) and then the other two channels should be connected. > But there is the problem. The the third party is disconnected too, when the operator hangs up the phone. > > I tried to catch the HangUpEvent or UnLinkEvent of the third channel and then redirect the third channel to the parking channel but it doesn´t work... > > Markus > > > > -----Ursprüngliche Nachricht----- > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet > Gesendet: Dienstag, 3. April 2007 09:19 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > Hi Markus > > If the channel is hung up when the operator hangs up the phone chances > are that the parking didn't work at all, because in our application as > soon as we redirect the other party to the parking, the operator line is > hung up by the system even before she puts the phone down (in fact the > phone is an headset with autoanswer in our case). > > In the dialplan I simply put this kind of thing: > > [CUSTOMPARK] > exten => *101,1,StopMusicOnHold() > exten => *101,2,StartMusicOnHold() > exten => *101,3,ParkSilent() > > Then I redirect the third party's leg to the extension > (CUSTOMPARK,*101,1) using AMI. > > Please note we don't use the parking extension declared in > features.general to avoid any interference with the stock parking > application, and that as a consequence the *101 is not automatically > inserted in the parkingext context by res_features. We use (and redirect > to) our own separate context. > > The only reason I see is that your parking extensions was not declared > in your context so the redirection simply fails as the dialplan can't > reach the extension. If you where using the stock parking application > and configuration, maybe you forgot to include the parkingext > (sub-)context in you incoming(zap ?) context ? > > > N.B: The stop/start MOH thing is here because with at least asterisk > 1.2.9 we had problem with the channel crashing when coming from a queue > where MOH was already playing, unless we used mpg123. That had something > to do with codec translation changes on the fly iirc; > > Gaetan > > Markus Floegel wrote: > >> Hi, >> >> I tried something like this before, but i had the problem that the third party was disconnected when the operator hangs up the phone. >> So i can´t redirect the channel of the third party to the parking channel. I alway gets an HangupEvent and the phone of the third party was disconnected. >> How do you implement this? >> >> thanks >> >> >> >> >> ________________________________ >> >> Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet >> Gesendet: Montag, 2. April 2007 11:02 >> An: ast...@li... >> Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API >> >> >> Hi, >> >> We managed to implement this using (well, abusing) the parking application. That way we have no meetme running for each attended call, the channels are always simply bridged. >> >> - We redirect the leg of the caller to the parking extension. The operator leg is then dropped and the caller has MOH. In return we get the parking position of the call. >> - We then call the other party, announce the call. >> - If he accepts the call, we redirect his current leg to the parking lot extension the call is parked at (and the operator leg is dropped once again). The other party gets the right tone just before the channels are bridged by the parking application. >> - If he refuses the call we redirect our own leg to the parking extension so we get the caller back. >> >> The only problem is that with the stock Park application, the caller is announced it's own parking lot extension so we made a tiny ast application, called "ParkSilent". That's just a copy&paste of the stock parking main function with second party channel set to null instead of the caller channel. This can be easily compiled against at least 1.2.9 - 1.2.16 without recompiling the whole asterisk (in fact it's even binary compatible as it links dynamically against the stock parking). >> >> Regards >> >> Gaetan >> >> >> >> >> Markus Floegel wrote: >> >> I think such a Action in the Manager will be very good. >> The workaround with the meetMe Application is not so fine, i think. >> >> Markus >> >> >> >> -----Ursprüngliche Nachricht----- >> Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Stefan Reuter >> Gesendet: Freitag, 30. März 2007 10:25 >> An: ast...@li... >> Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API >> >> Markus Floegel wrote: >> >> >> I implemented the same now im my project. >> But is there no way to use the internal 'attended transfer' from Asterisk? >> >> >> >> I've seen this question quite a few times now without any suitable answer. >> Maybe its time to implement a Manager Action for this feature and submit >> it as a patch to Asterisk? >> Anybody want's to take this? >> >> I would be happy to support that new action in Asterisk-Java. >> >> =Stefan >> >> >> >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to share your >> opinions on IT & business topics through brief surveys-and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> _______________________________________________ >> Asterisk-java-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >> >> >> > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > |
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From: Markus F. <mar...@ts...> - 2007-04-03 07:43:56
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Hi Gaetan, Thanks for the answer! >> the operator line is >> hung up by the system even before she puts the phone down (in fact = the >> phone is an headset with autoanswer in our case). In my application, the operator manualy hangs up the phone (using = hardware) and then the other two channels should be connected. But there is the problem. The the third party is disconnected too, when = the operator hangs up the phone. I tried to catch the HangUpEvent or UnLinkEvent of the third channel and = then redirect the third channel to the parking channel but it doesn=B4t = work... Markus =20 -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Dienstag, 3. April 2007 09:19 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API Hi Markus If the channel is hung up when the operator hangs up the phone chances are that the parking didn't work at all, because in our application as soon as we redirect the other party to the parking, the operator line is hung up by the system even before she puts the phone down (in fact the phone is an headset with autoanswer in our case). In the dialplan I simply put this kind of thing: [CUSTOMPARK] exten =3D> *101,1,StopMusicOnHold() exten =3D> *101,2,StartMusicOnHold() exten =3D> *101,3,ParkSilent() Then I redirect the third party's leg to the extension (CUSTOMPARK,*101,1) using AMI. Please note we don't use the parking extension declared in features.general to avoid any interference with the stock parking application, and that as a consequence the *101 is not automatically inserted in the parkingext context by res_features. We use (and redirect to) our own separate context. The only reason I see is that your parking extensions was not declared in your context so the redirection simply fails as the dialplan can't reach the extension. If you where using the stock parking application and configuration, maybe you forgot to include the parkingext (sub-)context in you incoming(zap ?) context ? N.B: The stop/start MOH thing is here because with at least asterisk 1.2.9 we had problem with the channel crashing when coming from a queue where MOH was already playing, unless we used mpg123. That had something to do with codec translation changes on the fly iirc; Gaetan Markus Floegel wrote: > Hi, > > I tried something like this before, but i had the problem that the = third party was disconnected when the operator hangs up the phone. > So i can=B4t redirect the channel of the third party to the parking = channel. I alway gets an HangupEvent and the phone of the third party = was disconnected. > How do you implement this? > > thanks > > > =20 > > ________________________________ > > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet > Gesendet: Montag, 2. April 2007 11:02 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > > Hi, > > We managed to implement this using (well, abusing) the parking = application. That way we have no meetme running for each attended call, = the channels are always simply bridged. > > - We redirect the leg of the caller to the parking extension. The = operator leg is then dropped and the caller has MOH. In return we get = the parking position of the call. > - We then call the other party, announce the call. > - If he accepts the call, we redirect his current leg to the parking = lot extension the call is parked at (and the operator leg is dropped = once again). The other party gets the right tone just before the = channels are bridged by the parking application. > - If he refuses the call we redirect our own leg to the parking = extension so we get the caller back. > > The only problem is that with the stock Park application, the caller = is announced it's own parking lot extension so we made a tiny ast = application, called "ParkSilent". That's just a copy&paste of the stock = parking main function with second party channel set to null instead of = the caller channel. This can be easily compiled against at least 1.2.9 - = 1.2.16 without recompiling the whole asterisk (in fact it's even binary = compatible as it links dynamically against the stock parking). > > Regards > > Gaetan > > > > > Markus Floegel wrote:=20 > > I think such a Action in the Manager will be very good. > The workaround with the meetMe Application is not so fine, i think. > =09 > Markus > =09 > =09 > =09 > -----Urspr=FCngliche Nachricht----- > Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Stefan Reuter > Gesendet: Freitag, 30. M=E4rz 2007 10:25 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager = API > =09 > Markus Floegel wrote: > =20 > > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from = Asterisk? > =20 > > =09 > I've seen this question quite a few times now without any suitable = answer. > Maybe its time to implement a Manager Action for this feature and = submit > it as a patch to Asterisk? > Anybody want's to take this? > =09 > I would be happy to support that new action in Asterisk-Java. > =09 > =3DStefan > =09 > =20 > > > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > =20 -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share = your opinions on IT & business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
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From: Gaetan M. <gm...@ea...> - 2007-04-03 07:18:54
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Hi Markus If the channel is hung up when the operator hangs up the phone chances are that the parking didn't work at all, because in our application as soon as we redirect the other party to the parking, the operator line is hung up by the system even before she puts the phone down (in fact the phone is an headset with autoanswer in our case). In the dialplan I simply put this kind of thing: [CUSTOMPARK] exten => *101,1,StopMusicOnHold() exten => *101,2,StartMusicOnHold() exten => *101,3,ParkSilent() Then I redirect the third party's leg to the extension (CUSTOMPARK,*101,1) using AMI. Please note we don't use the parking extension declared in features.general to avoid any interference with the stock parking application, and that as a consequence the *101 is not automatically inserted in the parkingext context by res_features. We use (and redirect to) our own separate context. The only reason I see is that your parking extensions was not declared in your context so the redirection simply fails as the dialplan can't reach the extension. If you where using the stock parking application and configuration, maybe you forgot to include the parkingext (sub-)context in you incoming(zap ?) context ? N.B: The stop/start MOH thing is here because with at least asterisk 1.2.9 we had problem with the channel crashing when coming from a queue where MOH was already playing, unless we used mpg123. That had something to do with codec translation changes on the fly iirc; Gaetan Markus Floegel wrote: > Hi, > > I tried something like this before, but i had the problem that the third party was disconnected when the operator hangs up the phone. > So i can´t redirect the channel of the third party to the parking channel. I alway gets an HangupEvent and the phone of the third party was disconnected. > How do you implement this? > > thanks > > > > > ________________________________ > > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Gaetan Minet > Gesendet: Montag, 2. April 2007 11:02 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > > Hi, > > We managed to implement this using (well, abusing) the parking application. That way we have no meetme running for each attended call, the channels are always simply bridged. > > - We redirect the leg of the caller to the parking extension. The operator leg is then dropped and the caller has MOH. In return we get the parking position of the call. > - We then call the other party, announce the call. > - If he accepts the call, we redirect his current leg to the parking lot extension the call is parked at (and the operator leg is dropped once again). The other party gets the right tone just before the channels are bridged by the parking application. > - If he refuses the call we redirect our own leg to the parking extension so we get the caller back. > > The only problem is that with the stock Park application, the caller is announced it's own parking lot extension so we made a tiny ast application, called "ParkSilent". That's just a copy&paste of the stock parking main function with second party channel set to null instead of the caller channel. This can be easily compiled against at least 1.2.9 - 1.2.16 without recompiling the whole asterisk (in fact it's even binary compatible as it links dynamically against the stock parking). > > Regards > > Gaetan > > > > > Markus Floegel wrote: > > I think such a Action in the Manager will be very good. > The workaround with the meetMe Application is not so fine, i think. > > Markus > > > > -----Ursprüngliche Nachricht----- > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Stefan Reuter > Gesendet: Freitag, 30. März 2007 10:25 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > Markus Floegel wrote: > > > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from Asterisk? > > > > I've seen this question quite a few times now without any suitable answer. > Maybe its time to implement a Manager Action for this feature and submit > it as a patch to Asterisk? > Anybody want's to take this? > > I would be happy to support that new action in Asterisk-Java. > > =Stefan > > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > |
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From: Markus F. <mar...@ts...> - 2007-04-03 06:44:52
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Hi, I tried something like this before, but i had the problem that the third = party was disconnected when the operator hangs up the phone. So i can=B4t redirect the channel of the third party to the parking = channel. I alway gets an HangupEvent and the phone of the third party = was disconnected. How do you implement this? thanks =20 ________________________________ Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Gaetan Minet Gesendet: Montag, 2. April 2007 11:02 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API Hi, We managed to implement this using (well, abusing) the parking = application. That way we have no meetme running for each attended call, = the channels are always simply bridged. - We redirect the leg of the caller to the parking extension. The = operator leg is then dropped and the caller has MOH. In return we get = the parking position of the call. - We then call the other party, announce the call. - If he accepts the call, we redirect his current leg to the parking lot = extension the call is parked at (and the operator leg is dropped once = again). The other party gets the right tone just before the channels are = bridged by the parking application. - If he refuses the call we redirect our own leg to the parking = extension so we get the caller back. The only problem is that with the stock Park application, the caller is = announced it's own parking lot extension so we made a tiny ast = application, called "ParkSilent". That's just a copy&paste of the stock = parking main function with second party channel set to null instead of = the caller channel. This can be easily compiled against at least 1.2.9 - = 1.2.16 without recompiling the whole asterisk (in fact it's even binary = compatible as it links dynamically against the stock parking). Regards Gaetan Markus Floegel wrote:=20 I think such a Action in the Manager will be very good. The workaround with the meetMe Application is not so fine, i think. =09 Markus =09 =09 =09 -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Stefan Reuter Gesendet: Freitag, 30. M=E4rz 2007 10:25 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API =09 Markus Floegel wrote: =20 I implemented the same now im my project. But is there no way to use the internal 'attended transfer' from = Asterisk? =20 =09 I've seen this question quite a few times now without any suitable = answer. Maybe its time to implement a Manager Action for this feature and = submit it as a patch to Asterisk? Anybody want's to take this? =09 I would be happy to support that new action in Asterisk-Java. =09 =3DStefan =09 =20 |
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From: Michael Y. <mic...@i9...> - 2007-04-02 15:33:06
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Before the Live API was introduced I would simply create that unknown channel in my model when I would get the rename event and copy the information from its relative channel. So in your trace when you would get the event: Rename: old: SIPPeer/SIP/1310-0820eee0 new: SIP/1310-0820eee0 I would create the channel SIPPeer/SIP/1310-0820eee0 then copy the state from SIP/1310-0820eee0<MASQ> and finally perform the rename event. Another more appropriate method may be to fire a StatusAction to the manager and look for that channel.|| -Mike Stefan Reuter wrote: > Michael Yara wrote: > >> I am using 1.2 >> > > It seems this problem relates to the following bug in Asterisk: > http://bugs.digium.com/view.php?id=7625 > We get a rename event for a channel that has not been announced before. > > the whole flow of events is more than weird when doing a blind transfer > to a parking lot. > > I've visualized a trace for that scenario at > http://www.reucon.com/~srt/trace.html > > Do you have an idea on how we could handle such a case? > > =Stefan > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
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From: Gaetan M. <gm...@ea...> - 2007-04-02 09:02:35
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Hi, We managed to implement this using (well, abusing) the parking application. That way we have no meetme running for each attended call, the channels are always simply bridged. - We redirect the leg of the caller to the parking extension. The operator leg is then dropped and the caller has MOH. In return we get the parking position of the call. - We then call the other party, announce the call. - If he accepts the call, we redirect his current leg to the parking lot extension the call is parked at (and the operator leg is dropped once again). The other party gets the right tone just before the channels are bridged by the parking application. - If he refuses the call we redirect our own leg to the parking extension so we get the caller back. The only problem is that with the stock Park application, the caller is announced it's own parking lot extension so we made a tiny ast application, called "ParkSilent". That's just a copy&paste of the stock parking main function with second party channel set to null instead of the caller channel. This can be easily compiled against at least 1.2.9 - 1.2.16 without recompiling the whole asterisk (in fact it's even binary compatible as it links dynamically against the stock parking). Regards Gaetan Markus Floegel wrote: > I think such a Action in the Manager will be very good. > The workaround with the meetMe Application is not so fine, i think. > > Markus > > > > -----Ursprüngliche Nachricht----- > Von: ast...@li... [mailto:ast...@li...] Im Auftrag von Stefan Reuter > Gesendet: Freitag, 30. März 2007 10:25 > An: ast...@li... > Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API > > Markus Floegel wrote: > >> I implemented the same now im my project. >> But is there no way to use the internal 'attended transfer' from Asterisk? >> > > I've seen this question quite a few times now without any suitable answer. > Maybe its time to implement a Manager Action for this feature and submit > it as a patch to Asterisk? > Anybody want's to take this? > > I would be happy to support that new action in Asterisk-Java. > > =Stefan > > |
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From: Stefan R. <ste...@re...> - 2007-04-01 19:30:28
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Michael Yara wrote: > I am using 1.2 It seems this problem relates to the following bug in Asterisk: http://bugs.digium.com/view.php?id=3D7625 We get a rename event for a channel that has not been announced before. the whole flow of events is more than weird when doing a blind transfer to a parking lot. I've visualized a trace for that scenario at http://www.reucon.com/~srt/trace.html Do you have an idea on how we could handle such a case? =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: Michael Y. <mic...@i9...> - 2007-03-30 15:52:25
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I am using 1.2 Stefan Reuter wrote: > Michael Yara wrote: > >> When i park a call via a blind transfer i noticed that a lot of events >> don't fire. I checked the log and it seems that asterisk-java is >> reporting errors about channel SIP/42242-09060eb8 and that it does not >> exist and failing to fire specific events, but the call parks fine. >> > > Thanks for you feedback. I will have a look at it this weekend and > report back. > Which version of Asterisk are you using? > > =Stefan > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
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From: Markus F. <mar...@ts...> - 2007-03-30 10:13:34
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I think such a Action in the Manager will be very good. The workaround with the meetMe Application is not so fine, i think. Markus -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Stefan Reuter Gesendet: Freitag, 30. M=E4rz 2007 10:25 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API Markus Floegel wrote: > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from = Asterisk? I've seen this question quite a few times now without any suitable = answer. Maybe its time to implement a Manager Action for this feature and submit it as a patch to Asterisk? Anybody want's to take this? I would be happy to support that new action in Asterisk-Java. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: Stefan R. <ste...@re...> - 2007-03-30 08:25:05
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Markus Floegel wrote: > I implemented the same now im my project. > But is there no way to use the internal 'attended transfer' from Asteri= sk? I've seen this question quite a few times now without any suitable answer= =2E Maybe its time to implement a Manager Action for this feature and submit it as a patch to Asterisk? Anybody want's to take this? I would be happy to support that new action in Asterisk-Java. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: Stefan R. <ste...@re...> - 2007-03-30 08:22:17
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Michael Yara wrote: > When i park a call via a blind transfer i noticed that a lot of events = > don't fire. I checked the log and it seems that asterisk-java is=20 > reporting errors about channel SIP/42242-09060eb8 and that it does not = > exist and failing to fire specific events, but the call parks fine. Thanks for you feedback. I will have a look at it this weekend and report back. Which version of Asterisk are you using? =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: ste...@re... Jabber: ste...@re... Steuernummern 215/5140/1791 USt-IdNr. DE220701760 |
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From: Markus F. <mar...@ts...> - 2007-03-30 06:24:15
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Thanks for the answer!=20 I implemented the same now im my project. But is there no way to use the internal 'attended transfer' from = Asterisk? Markus -----Urspr=FCngliche Nachricht----- Von: ast...@li... = [mailto:ast...@li...] Im Auftrag = von Johannes Boesl Gesendet: Donnerstag, 29. M=E4rz 2007 12:17 An: ast...@li... Betreff: Re: [Asterisk-java-users] Attended Transfer using Manager API Hi there, I did implement transfer based on conference. That means you put your=20 first call onhold, call a second party, join everybody in a conference=20 and drop yourself. It's available in the cvs of the GJTapi-project=20 (http://sourceforge.net/projects/gjtapi). Kind regards, Johannes Boesl > Hello, > > Did anyone implemented a attended transfer using the Manager API? > > I have a WebApplikation wich should initiate an attended transfer. > The blind transfer using the RedirectAction works fine. > > Some about my situation: > I know the Channel of the Phone that will do the transfer.=20 > > The attended transfer is configured in the features.conf. > On the phone itself it works fine. > I did some tests with the PlayDTMF, but it doesn=B4t work. I still her = the tone in the phone, but there is no reaktion from asterisk. > > Markus > > = -------------------------------------------------------------------------= > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to = share your > opinions on IT & business topics through brief surveys-and earn cash > = http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > =20 -------------------------------------------------------------------------= Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share = your opinions on IT & business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDEV _______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
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From: Johannes B. <j....@ad...> - 2007-03-29 10:17:04
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Hi there, I did implement transfer based on conference. That means you put your first call onhold, call a second party, join everybody in a conference and drop yourself. It's available in the cvs of the GJTapi-project (http://sourceforge.net/projects/gjtapi). Kind regards, Johannes Boesl > Hello, > > Did anyone implemented a attended transfer using the Manager API? > > I have a WebApplikation wich should initiate an attended transfer. > The blind transfer using the RedirectAction works fine. > > Some about my situation: > I know the Channel of the Phone that will do the transfer. > > The attended transfer is configured in the features.conf. > On the phone itself it works fine. > I did some tests with the PlayDTMF, but it doesn´t work. I still her the tone in the phone, but there is no reaktion from asterisk. > > Markus > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys-and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
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From: Markus F. <mar...@ts...> - 2007-03-29 10:11:58
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Hello, Did anyone implemented a attended transfer using the Manager API? I have a WebApplikation wich should initiate an attended transfer. The blind transfer using the RedirectAction works fine. Some about my situation: I know the Channel of the Phone that will do the transfer.=20 The attended transfer is configured in the features.conf. On the phone itself it works fine. I did some tests with the PlayDTMF, but it doesn=B4t work. I still her = the tone in the phone, but there is no reaktion from asterisk. Markus |
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