Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR
On Wed, Dec 17, 2008 at 2:16 PM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. you mean second option is not acceptable, don't you ? Sorry that is my mistake and you are right. I wanna use one channel and only while call is inside the IVR. Bye, Zsolt ___ -- 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] Alcatel OXE + Asterisk as external IVR
2008/12/18 Artifex Maximus artife...@gmail.com On Wed, Dec 17, 2008 at 2:16 PM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. you mean second option is not acceptable, don't you ? Sorry that is my mistake and you are right. I wanna use one channel and only while call is inside the IVR. No problem ! By the way, have you found any doc describing Asterisk Q.SIG features ? Bye, Zsolt ___ -- 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 ___ -- 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] Alcatel OXE + Asterisk as external IVR
2008/12/17 Artifex Maximus artife...@gmail.com On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. Thanks for your answer! You are right and first option what I am looking for. I have asked support staff and sending back DTMF on open channel does not help. True ! I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. Unfortunately I am not expert on this topic as well but second option is not good for us. The question is how good Asterisk's Q.SIG implementation for this task. That's the question ! Maybe someone else could help on this as I don't have much experience to share. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. you mean second option is not acceptable, don't you ? Bye, Zsolt ___ -- 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 ___ -- 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] Alcatel OXE + Asterisk as external IVR
On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. Thanks for your answer! You are right and first option what I am looking for. I have asked support staff and sending back DTMF on open channel does not help. I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. Unfortunately I am not expert on this topic as well but second option is not good for us. The question is how good Asterisk's Q.SIG implementation for this task. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. Bye, Zsolt ___ -- 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] Alcatel OXE + Asterisk as external IVR
2008/12/17 Artifex Maximus artife...@gmail.com Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR transfer back to specified Pilot in OXE with Dial (using two channels) 6. Asterisk hangup (free both channels) 7. OXE connect the PSTN incoming line with Pilot as extension transfer does I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? I am using Asterisk 1.6.0.3-rc1 with dahdi-*-2.1.0 on Ubuntu Server 8.10. Thanks, Zsolt Hi, What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. ___ -- 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 ___ -- 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] Alcatel OXE + Asterisk as external IVR
Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR transfer back to specified Pilot in OXE with Dial (using two channels) 6. Asterisk hangup (free both channels) 7. OXE connect the PSTN incoming line with Pilot as extension transfer does I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? I am using Asterisk 1.6.0.3-rc1 with dahdi-*-2.1.0 on Ubuntu Server 8.10. Thanks, Zsolt ___ -- 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
