Thread: [Asterisk-java-users] RedirectAction to MeetMe
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From: Johannes B. <j....@ad...> - 2006-10-12 10:10:27
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Hello, =20 I'm currently implementing 'join' for the GJTapi-Asterisk-Provider. This = enables conference and transfer for GJTapi applications using that = provider. It's based on Asterisk-Java's manager interface. But I'm = running into problems ... maybe one can help me here. =20 A scenario is:=20 someone calls me on my SIP-phone from 'outside' over Zap. I place that = person on hold and call another party. That done my application knows = there are two active calls with four different channels.=20 What I want to do next is to 'join' those two calls by using MeetMe and = the RedirectAction. Asterisk has an extension called 'custom-conf': = 'exten =3D> s,1,MeetMe(${EXTEN},dpqM)'. So I try to redirect one of my = SIP-Channels and the other two channels connected to the other parties = to a MeetMe-Conference: RedirectAction redAct =3D new RedirectAction(channel1, "custom-conf", = "222", new Integer(1)); managerConnection.sendAction(redAct); And the same for the other channels. The result is disconnection of all = parties. The response.getMessage() function keeps telling me "Channel = does not exist: net.sf.asterisk.manager.Channel: = id=3D'asterisk-6535-1160646952.173'; name=3D'Zap/1-1'; = callerId=3D'96....." for the external Channel no matter whether it's a = Sip or a Zap channel. Calling asteriskManager.getChannels() I can see = that it exists. Asterisk tells me " =3D=3D Spawn extension = (custom-conf, 222, 0) exited non-zero on 'Zap/1-1' in macro 'dial'" = followed by " -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1'" =20 Any ideas? Maybe I should mention I'm using Asterisk-Java 0.2 because = I'm forced to use 1.4 Java. =20 Thanks, Johannes Boesl |
From: Johannes B. <j....@ad...> - 2006-10-16 09:10:30
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Hello, =20 I'm currently implementing 'join' for the GJTapi-Asterisk-Provider. This = enables conference and transfer for GJTapi applications using that = provider. It's based on Asterisk-Java's manager interface. But I'm = running into problems ... maybe one can help me here. =20 A scenario is:=20 someone calls me on my SIP-phone from 'outside' over Zap. I place that = person on hold and call another party. That done my application knows = there are two active calls with four different channels.=20 What I want to do next is to 'join' those two calls by using MeetMe and = the RedirectAction. Asterisk has an extension called 'custom-conf': = 'exten =3D> s,1,MeetMe(${EXTEN},dpqM)'. So I try to redirect one of my = SIP-Channels and the other two channels connected to the other parties = to a MeetMe-Conference: RedirectAction redAct =3D new RedirectAction(channel1, "custom-conf", = "222", new Integer(1)); managerConnection.sendAction(redAct); And the same for the other channels. The result is disconnection of all = parties. The response.getMessage() function keeps telling me "Channel = does not exist: net.sf.asterisk.manager.Channel: = id=3D'asterisk-6535-1160646952.173'; name=3D'Zap/1-1'; = callerId=3D'96....." for the external Channel no matter whether it's a = Sip or a Zap channel. Calling asteriskManager.getChannels() I can see = that it exists. Asterisk tells me " =3D=3D Spawn extension = (custom-conf, 222, 0) exited non-zero on 'Zap/1-1' in macro 'dial'" = followed by " -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1'" =20 Any ideas? Maybe I should mention I'm using Asterisk-Java 0.2 because = I'm forced to use 1.4 Java. =20 Thanks, Johannes Boesl |
From: Stefan R. <ste...@re...> - 2006-10-14 17:34:03
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do you use a bristuffed version of asterisk? There were (is?) a bug that kpj never fixed though I sent him my patch several time. Its at http://www.reucon.net/~srt/bristuff_q_redirect.patch= =3DStefan Johannes Boesl wrote: > Hello, > =20 > I'm currently implementing 'join' for the GJTapi-Asterisk-Provider. Thi= s > enables conference and transfer for GJTapi applications using that > provider. It's based on Asterisk-Java's manager interface. But I'm > running into problems ... maybe one can help me here. > =20 > A scenario is: > someone calls me on my SIP-phone from 'outside' over Zap. I place that > person on hold and call another party. That done my application knows > there are two active calls with four different channels. > What I want to do next is to 'join' those two calls by using MeetMe and= > the RedirectAction. Asterisk has an extension called 'custom-conf': > 'exten =3D> s,1,MeetMe(${EXTEN},dpqM)'. So I try to redirect one of my > SIP-Channels and the other two channels connected to the other parties > to a MeetMe-Conference: > RedirectAction redAct =3D new RedirectAction(channel1, "custom-conf", > "222", new Integer(1)); > managerConnection.sendAction(redAct); > And the same for the other channels. The result is disconnection of all= > parties. The response.getMessage() function keeps telling me "Channel > does not exist: net.sf.asterisk.manager.Channel: > id=3D'asterisk-6535-1160646952.173'; name=3D'Zap/1-1'; callerId=3D'96..= =2E.." > for the external Channel no matter whether it's a Sip or a Zap channel.= > Calling asteriskManager.getChannels() I can see that it exists. Asteris= k > tells me " =3D=3D Spawn extension (custom-conf, 222, 0) exited non-zer= o on > 'Zap/1-1' in macro 'dial'" followed by " -- Stopped music on hold on= > Zap/1-1 > -- Hungup 'Zap/1-1'" > =20 > Any ideas? Maybe I should mention I'm using Asterisk-Java 0.2 because > I'm forced to use 1.4 Java. > =20 > Thanks, > Johannes Boesl >=20 >=20 > -----------------------------------------------------------------------= - >=20 > -----------------------------------------------------------------------= -- > Using Tomcat but need to do more? Need to support web services, securit= y? > Get stuff done quickly with pre-integrated technology to make your job = easier > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geron= imo > http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat= =3D121642 >=20 >=20 > -----------------------------------------------------------------------= - >=20 > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users --=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... |