RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
On 12/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. Some options of the Dial command force * to stay in the media path, like t (to let user transfer by hitting #). So you could just put one of thos options in your dial string hth ___ --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
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Yeah, makes sense now that I think about it a little more. Guess you will have to prefix your exten so that the dial string with the H is used and dial that prefix when you know or think that you may have to record a call. This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
I have a related issue. I have everything set up correctly so that I CAN use live recording (Press *1 to start and stop recording.) When I press *1, the console indicates user pressed *1 to start recording. I also hear the beep and an audio file is created. The problem is that the audio file IS NOTHING BUT SILENCE. It is the correct length, but only contains silence. Any ideas??? -N On Dec 8, 2005, at 8:49 AM, Steve Totaro wrote: Yeah, makes sense now that I think about it a little more. Guess you will have to prefix your exten so that the dial string with the H is used and dial that prefix when you know or think that you may have to record a call. This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --Bandwidth and
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Hi! This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. You might even have another option: DTMF via SIP INFO Quote from asterisk-devel two days ago: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/14693 Cheers, Philipp Wolfgang S. Rupprecht wrote: I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? This is certainly possible, but Asterisk currently assumes that if it is not in the media path, it also won't be able to receive DTMF frames. However, if you are using SIP INFO for DTMF signaling, then it should 'just work', since when Asterisk sees the appropriate DTMF frames it will cause the bridge to 'break' and bring the media path back. ___ --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
RE: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Do you have canreinvite=yes? If you do change it to no. If that works then read the rest of this thread for options if you do not want all streams to through asterisk. Thanks, Steve I have a related issue. I have everything set up correctly so that I CAN use live recording (Press *1 to start and stop recording.) When I press *1, the console indicates user pressed *1 to start recording. I also hear the beep and an audio file is created. The problem is that the audio file IS NOTHING BUT SILENCE. It is the correct length, but only contains silence. Any ideas??? -N On Dec 8, 2005, at 8:49 AM, Steve Totaro wrote: Yeah, makes sense now that I think about it a little more. Guess you will have to prefix your exten so that the dial string with the H is used and dial that prefix when you know or think that you may have to record a call. This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. I'll give it a try. Thanks, Waldo On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote: There may be a better way but off the top of my head this idea jumped out. It assumes that you know prior to making the call that you need to record it and that you have phones capable of multiple lines. Setup a second line with a different entry in sip.conf with canreinvite=no and use that line to make your calls. Other than that I see reference on the wiki to an H option in dial but have never used it. I think you will still need to know prior to dialing whether you will want to record the call or not so you can dial the exten that uses the H option. If you get this to work, please post your results back to this thread. Re: Re: H option by flobi on Monday 25 of July, 2005 [10:43:46] why not just set canreinvite=yes and on the calls where you don't want reinvite use the H option (if it actually does disable reinvite) or the T or t which also disable reinvite. 7960G Seems to need canreinvite=no as well. by Anonymous on Friday 29 of October, 2004 [22:22:43] Running P0S3-07-2-00. Re: H option by Anonymous on Monday 26 of July, 2004 [10:10:07] (:confused:) Hmm... Now I started to wonder, if it's somehow possible to override the canreinvite=no setting on per call basis. Anyone? H option by Anonymous on Saturday 10 of July, 2004 [04:15:13] Asterisk will not reinvite if the H option is used in the Dial command. http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Thanks, Steve I understand. But because the majority of calls are not to be recorded, I don't have a need to keep Asterisk in the media path all the time. That's why I'm wondering if you could dynamically keep it in the media path or not. - Waldo On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote: Well, then set canreinvite=no If that's the case, is it possible to override the canreinvite attribute of a SIP peer in extensions.conf before a call is made or answered by that peer? - Waldo On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote: Is there a way to optionally keep asterisk in the media path in order to record calls using the Monitor command? For example, if I have a SIP peer that is defined with canreinvite=yes, this means that if possible, Asterisk will not be in the media path. However, what happens if the user presses something like *1 (defined in features.conf) to record the call? Will the call be forced to go through Asterisk automatically? Thanks, Waldo I could be wrong but I am pretty sure that once the asterisk is out of the media path then features like *1 will not work since asterisk is not able to listen for it. Thanks, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing
Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite
Has anyone confirmed this? It sounds like an interesting theory. - Waldo On Dec 8, 2005, at 12:46 PM, Philipp von Klitzing wrote: Hi! This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. You might even have another option: DTMF via SIP INFO Quote from asterisk-devel two days ago: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/14693 Cheers, Philipp Wolfgang S. Rupprecht wrote: I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? This is certainly possible, but Asterisk currently assumes that if it is not in the media path, it also won't be able to receive DTMF frames. However, if you are using SIP INFO for DTMF signaling, then it should 'just work', since when Asterisk sees the appropriate DTMF frames it will cause the bridge to 'break' and bring the media path back. ___ --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
