Re: [asterisk-users] meetme and dtmf
I don't get what the 'F' option is for. Its not proper to exit a context and then reenter the conference as admin Isn't there any other way to do actions such as kick/mute/unmute users by admin dtmf trigger? On Fri, Jun 1, 2012 at 3:47 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 31 May 2012, Daniel Knoll wrote: is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. I'm just a 1.2 Luddite, but... You can use the meetme() 'X' option to jump out of the meetme and into another context. I use this to allow conference administrators to mute, un-mute, or kick users. The first digit jumps out of the meetme and into another context where I read additional digits (the user index) and then call an AGI (meetmeadmin-by-index) before returning the admin to the conference. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme and dtmf
Hi Group, is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. Thanks for help Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme and dtmf
On Thu, 31 May 2012, Daniel Knoll wrote: is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. I'm just a 1.2 Luddite, but... You can use the meetme() 'X' option to jump out of the meetme and into another context. I use this to allow conference administrators to mute, un-mute, or kick users. The first digit jumps out of the meetme and into another context where I read additional digits (the user index) and then call an AGI (meetmeadmin-by-index) before returning the admin to the conference. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Imran Ahmed wrote: may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) Thank you very much. I tried sjphone setting clinet and asterisk as above and it seems to work. I will test it better in the next hours. I had a look at meetme.c and i found a portion of code that manage dtmf if ((f-frametype == AST_FRAME_DTMF) (confflags CONFFLAG_EXIT_CONTEXT)) { .. .. - I think this part manage the case of meetme application is called with p, X or s option, but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. Sorry if my bad english make me not very clear. Anyway, thank you very much to all for your help. Accursio Avona ___ --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] meetme and dtmf
Accursio Avona wrote: but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. MeetMe is not designed to pass DTMF through between the parties in any case. It may happen if you use inband DTMF and don't have Asterisk actually paying attention to DTMF for any reason, but it's not intended to work that way. ___ --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] meetme and dtmf
Kevin P. Fleming wrote: but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. MeetMe is not designed to pass DTMF through between the parties in any case. It may happen if you use inband DTMF and don't have Asterisk actually paying attention to DTMF for any reason, but it's not intended to work that way. This means that if i'd like to use iax2 protocol (i need to integrate, into a propietary crm, calling features though asterisk, and i thougth to use iaxclient dll) i can't pass DTMF through between the parties? If so is it possible to modify meetme.c to avoid this behaviour? or i must use sip protocol. Thank's Accursio Avona ___ --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] meetme and dtmf
Imran Ahmed wrote: On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2 is defined to always send out-of-band DTMF. At best, if the receiving IAX2 system is just passing the audio along to another protocol that does support inband DTMF, then sending it in the audio stream would work. If the application receiving the DTMF is on the other IAX2 end, though (like MeetMe in this case), then it will never 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF. I agree, but the other ends of the conference were zap channels in this case, at least that is what I figured by the first email. Maybe if a paint better my scenario it would help the discussion. Step 1: A IAX client make a call executing the following command Dial(ZAP/g1/${EXTEN}) If aswered this call is tranfered to a conference room. Step 2: The IAX client make a second call executing again Dial(ZAP/g1/${EXTEN}) an IVR answer this call and the IAX client have to send some DTMF stil now everything works very well. At this point call is transfered to the previous conference room and The IAX client reach the conference too. Step 3 The Iax client heve to send some other DTMF to the IVR. NOW THE IVR DOES NOT HEAR DTMF SENDED BY THE IAX CLIENT, EVEN IF IT CAN HEAR DTMF SENDED BY THE FIRST ZAP CHANNEL. Hoping to be clear enough thank yuo very much for any help or suggestion. Accursio Avona ___ --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] meetme and dtmf
Accursio Avona wrote: Step 2: The IAX client make a second call executing again Dial(ZAP/g1/${EXTEN}) an IVR answer this call and the IAX client have to send some DTMF stil now everything works very well. At this point call is transfered to the previous conference room and The IAX client reach the conference too. Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Kevin P. Fleming wrote: Accursio Avona wrote: Step 2: The IAX client make a second call executing again Dial(ZAP/g1/${EXTEN}) an IVR answer this call and the IAX client have to send some DTMF stil now everything works very well. At this point call is transfered to the previous conference room and The IAX client reach the conference too. Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. ___ --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] meetme and dtmf
Accursio Avona wrote: The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. Then I really don't understand at all... this is not functionality that I would call an 'IVR'. Can you show us the portions of the Asterisk dialplans that are involved here? ___ --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] meetme and dtmf
Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran ___ --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] meetme and dtmf
On Fri, February 3, 2006 0:44, Imran Ahmed said: Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran What I find strange is that the meetme IVR participant *does* hear DTMF from the ZAP channel, but not from the IAX2 channel... There shouldn't be a per channel difference in how dtmf is handled in meetme, right?... Do you know whether the IAX2 dtmf is intercepted by meetme and handled internally? If so you might be able to workaround by using SendDTMF() in your meetme dialplan... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance Regards Accursio Avona ___ --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] meetme and dtmf
On Wed, February 1, 2006 12:07, Accursio Avona said: Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
Francesco Peeters (Asterisk) wrote: On Wed, February 1, 2006 12:07, Accursio Avona said: Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX If so, wy the IVR does not hear the dtmf sended by the iax client and it hear that one sendee by the zap channel? Could it be a meetme problem? and if so what can i do? Thank yuo very much for any help. Accursio Avona ___ --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] meetme and dtmf
On Wed, February 1, 2006 15:04, Accursio Avona said: Francesco Peeters (Asterisk) wrote: SNIP AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX If so, wy the IVR does not hear the dtmf sended by the iax client and it hear that one sendee by the zap channel? Could it be a meetme problem? and if so what can i do? Thank yuo very much for any help. Accursio Avona Are you sure it *is* sending DTMF in the first place? (Just trying to find a logical place to start here...) I do not use meetme, but when I use idefisk, my (*) server recognizes the DTMF. Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE meetme? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
Francesco Peeters (Asterisk) wrote: Are you sure it *is* sending DTMF in the first place? (Just trying to find a logical place to start here...) I do not use meetme, but when I use idefisk, my (*) server recognizes the DTMF. Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE meetme? Yes it is, outside meetme everything works fine. Thank's for any help or suggestion. Accursio Avona ___ --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] meetme and dtmf
AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. The problem here is with the meetme application when dealing with non zaptel channels it does not have a mechanism to enable dtmf to pass through the conference unless dtmf is inband (i.e. part of the audio stream). The following are the solutions a) Use a SIP phone with inband dtmf (No guarantee this will work either) b) Modify meetme to broadcast dtmf to all channels in conference( All channels will work in this case). Imran ___ --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] meetme and dtmf
Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2 is defined to always send out-of-band DTMF. At best, if the receiving IAX2 system is just passing the audio along to another protocol that does support inband DTMF, then sending it in the audio stream would work. If the application receiving the DTMF is on the other IAX2 end, though (like MeetMe in this case), then it will never 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF. ___ --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] meetme and dtmf
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2 is defined to always send out-of-band DTMF. At best, if the receiving IAX2 system is just passing the audio along to another protocol that does support inband DTMF, then sending it in the audio stream would work. If the application receiving the DTMF is on the other IAX2 end, though (like MeetMe in this case), then it will never 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF. I agree, but the other ends of the conference were zap channels in this case, at least that is what I figured by the first email. Imran. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme and dtmf
Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos conference room. At this point iax client joins to the conference and talking to the first zap channel need to send dtmf to the IVR. Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. Is there someone that can help me? any suggestion i welcome. Best Regards Accursio Avona ___ --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] meetme and dtmf
Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme: Sending DTMF to other users in a conference
Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference
Hello, We wrote a small AGI script to do just this. We just drop it's exten into the conference room and pass it the digits you want played and it will play the audio files of DTMF digits to all participants in the meetme room. It works great for us and we've been using it for over 2 years now. MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference
The script is part of the astGUIclient package(http://astguiclient.sf.net) Here's a direct link to the agi script itself: http://astguiclient.sf.net/experimental_code/agi-dtmf.agi ; this is used for sending DTMF signals within conference calls ;sends the digits to be played in the callerID field ;sound files must be placed in /var/lib/asterisk/sounds exten = 8500998,1,Answer exten = 8500998,2,AGI(agi-dtmf.agi) exten = 8500998,3,Hangup I use a manager API Action call to trigger the agi: In this example 78600051 is the exten to silently enter the meetme conf Action: Originate Channel: local/[EMAIL PROTECTED] Context: demo Exten: 78600051 Priority: 1 Callerid: 123,,456 Hope this helps, MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi Matt, Do you mind sharing that AGI script and the exact procedure in detail with me. I would be very thankful to you. Thanks, ~Vamsi -- Forwarded message -- From: Matt Florell [EMAIL PROTECTED] Date: Nov 4, 2005 10:03 PM Subject: Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello, We wrote a small AGI script to do just this. We just drop it's exten into the conference room and pass it the digits you want played and it will play the audio files of DTMF digits to all participants in the meetme room. It works great for us and we've been using it for over 2 years now. MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users