Re: [asterisk-users] Inserting a wait in a sip dial
2010/1/12 Kevin P. Fleming kpflem...@digium.com ... 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). In general, dial-string feature codes like this are not used on 'intelligent signaling' channels like SIP and ISDN; there are nearly always other, proper, ways to get the desired effect. Where are those dialing options documented ? core show application Dial doesn't and I would consider this logical as awaited information somehow depends on channel type. -- _ -- 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] Inserting a wait in a sip dial
12 jan 2010 kl. 20.56 skrev David Gibbons: snip 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). /snip Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? So: 1. dial 12345 2. connect SIP provider to * extension 3. wait 2 seconds programmatically 3. inject 567 DTMF tones into channel to signal remote PBX of extension to dial I'm hoping there's another way to skin this cat. From show application dial D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. /O -- _ -- 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] Inserting a wait in a sip dial
12 jan 2010 kl. 19.47 skrev Danny Nicholas: Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... I have no solution, but can only say this: a 'w' in a SIP dialstring doesn't produce any wait protocol-wise. SIP is all enbloc signalling. The gateway from SIP to PSTN might have an implementation of old hayes-like commands and support w for inserting wait periods, but you will have to check the documentation for that gateway. /O -- _ -- 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] Inserting a wait in a sip dial
snip But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. /snip I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _ -- 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] Inserting a wait in a sip dial
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, January 12, 2010 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Inserting a wait in a sip dial snip But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. /snip I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _ -- 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] Inserting a wait in a sip dial
The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert -- _ -- 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] Inserting a wait in a sip dial
This doesn't work? Dial(SIP/*31#ww061234123412) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ev...@disruptor.nl Sent: Tuesday, January 12, 2010 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inserting a wait in a sip dial The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert -- _ -- 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] Inserting a wait in a sip dial
Ok my problem is solved now, it was easyer fixed by adding: Set(CALLERPRES()=unavailable) That did exactly the same as the *31# would have done. So for me the problem is solved. The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert Regards, Evert -- _ -- 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] Inserting a wait in a sip dial
snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? -- _ -- 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] Inserting a wait in a sip dial
David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). In general, dial-string feature codes like this are not used on 'intelligent signaling' channels like SIP and ISDN; there are nearly always other, proper, ways to get the desired effect. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Inserting a wait in a sip dial
snip 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). /snip Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? So: 1. dial 12345 2. connect SIP provider to * extension 3. wait 2 seconds programmatically 3. inject 567 DTMF tones into channel to signal remote PBX of extension to dial I'm hoping there's another way to skin this cat. -Dave -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
Dave-- I remember adding a feature a long time ago for snoms, to the source code, to send dtmf out for some button press on a snom phone, in the 'outward' direction, I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any rate, I was able to inject dtmf, but I had to do it in the source. AFAICT, there is no app that do this explicitly; and Murphy's Law would state that even if a dialplan app existed, it would not get run at the time you need to be run. So, if you found a workaround, and it works, it won't matter how pretty it is. Magic is Magic. And speaking of Murphy's Law: Enjoy it while it lasts, because, sure as death and taxes, someone will fix a bug somewhere, and you'll lose an undocumented feature ;) murf On Tue, Jan 12, 2010 at 2:31 PM, David Gibbons d...@videon-central.comwrote: snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
If you need to inject dtmf tones or sound into an existing channel you can use chanspy with option w. I play sound files using the AMI to originate a call to an extension that does chanspy on one leg and a playback on the other. I use channel variables to say which channel to play to and which sound file to play. SendDTMF or Playtones should be able to inject tones. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2010, at 4:19 PM, Steve Murphy wrote: Dave-- I remember adding a feature a long time ago for snoms, to the source code, to send dtmf out for some button press on a snom phone, in the 'outward' direction, I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any rate, I was able to inject dtmf, but I had to do it in the source. AFAICT, there is no app that do this explicitly; and Murphy's Law would state that even if a dialplan app existed, it would not get run at the time you need to be run. So, if you found a workaround, and it works, it won't matter how pretty it is. Magic is Magic. And speaking of Murphy's Law: Enjoy it while it lasts, because, sure as death and taxes, someone will fix a bug somewhere, and you'll lose an undocumented feature ;) murf On Tue, Jan 12, 2010 at 2:31 PM, David Gibbons d...@videon-central.com wrote: snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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] Inserting a wait in a sip dial
Kevin P. Fleming wrote: David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). hmm, I use 'w' on ISDN channels (libpri) to signal sending complete, like Dial(DAHDI/g1/0123456w). But I did not know that 'w' means actually 'wait'. Regards Hans -- _ -- 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
