Re: [asterisk-users] Orginate not working well with PSTN lines
On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com wrote: I know this thread is dead but: I do not believe this should go into the DAHDI kernel modules. I agree. It's just too bad Dahdi is unable to report how an outgoing call is doing: Still ringing, busy, answered. -- _ -- 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] Orginate not working well with PSTN lines
5 maj 2011 kl. 16.35 skrev Gilles: On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com wrote: I know this thread is dead but: I do not believe this should go into the DAHDI kernel modules. I agree. It's just too bad Dahdi is unable to report how an outgoing call is doing: Still ringing, busy, answered. Just to add to the confusion... I have a branch where I managed to get manager originate to handle early media. If we get 183 (sip) or progress in ISDN with media before the answer, a manager originate will start the bridge. We're using that to get the Telco messages when we dial out to connect to a meetme. Previously we just had failed calls, but now we can hear the Telco message saying something like Invalid number, please try again or Weasles have eaten your phone system In the SIP channel, I would like to send some sort of control message when we get 100 trying. This means that we at least have a connection to something, even if we don't know if we've reached the target endpoint. I don't know if there's a similar message in ISDN, PSTN or other channels. But that's another patch :-) /O -- _ -- 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] Orginate not working well with PSTN lines
On Fri, Apr 29, 2011 at 01:04:42AM +0200, Gilles wrote: On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . I guess it was a lot of work, and nobody bothered adding this to the Zaptel driver. I know this thread is dead but: I do not believe this should go into the DAHDI kernel modules. The only thing the kernel modules could possibly do if the ring tone is detected is queue an event on the channel for Asterisk to decide how to handle. Asterisk / chan_dahdi is already typically monitoring the channel for DTMF digits and looking for additional tones and patterns could be added there. Asterisk would potentially need to have the tonezones of all possible destinations loaded which would make this complex and resource hungry. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Orginate not working well with PSTN lines
I thank everyone, for their fruitfull informations. Regards, Ashik Ali On Fri, Apr 29, 2011 at 2:04 AM, Gilles codecompl...@free.fr wrote: On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . I guess it was a lot of work, and nobody bothered adding this to the Zaptel driver. -- _ -- 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 -- _ -- 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] Orginate not working well with PSTN lines
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . I guess it was a lot of work, and nobody bothered adding this to the Zaptel driver. -- _ -- 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] Orginate not working well with PSTN lines
Dear all, The problem here is that as soon as asterisk dialing on fxo lines it sets channel status as answered although the chennel is getting ring back tone from other party. Anyone can suggest me to solve this issue ? Thanks , Ashik On Tue, Apr 26, 2011 at 4:28 PM, Jim Dickenson dicken...@cfmc.com wrote: Originate successfully queued only means that the originate action was handed off to asterisk not that is was executed yet. There are other events, depending on which events you are reading, that tell you the call was answered and such. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote: Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890 Response: Success ActionID: ABC45678901234567890 Message: Originate successfully queued -- Remote UNIX connection disconnected Channel DAHDI/1-1 was answered. -- Executing [1234@outbound-ivr:1] SayDigits(DAHDI/1-1, 1234) in new stack -- DAHDI/1-1 Playing 'digits/1.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/2.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/3.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/4.gsm' (language 'en') -- Executing [1234@outbound-ivr:2] Playback(DAHDI/1-1, demo-congrats) in new stack -- DAHDI/1-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [1234@outbound-ivr:3] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' Thanks Regards, Ashik -- _ -- 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 -- _ -- 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 -- _ -- 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] Orginate not working well with PSTN lines
On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: The problem here is that as soon as asterisk dialing on fxo lines it sets channel status as answered although the chennel is getting ring back tone from other party. Anyone can suggest me to solve this issue ? The only solution I know is to have Asterisk play a message in a loop for eg. 1mn, prompting the callee to hit a key to let the server know that the call was 1) answered 2) by a human being. -- _ -- 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] Orginate not working well with PSTN lines
Thanks for your solution. Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . Does anybody successed; to make asterisk to detect ring back tone from PSTN telco ? Thanks, Ashik On Wed, Apr 27, 2011 at 12:44 PM, Gilles codecompl...@free.fr wrote: On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali beaasteriskg...@gmail.com wrote: The problem here is that as soon as asterisk dialing on fxo lines it sets channel status as answered although the chennel is getting ring back tone from other party. Anyone can suggest me to solve this issue ? The only solution I know is to have Asterisk play a message in a loop for eg. 1mn, prompting the callee to hit a key to let the server know that the call was 1) answered 2) by a human being. -- _ -- 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 -- _ -- 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] Orginate not working well with PSTN lines
When dialing is finished on an analog FXO Asterisk considers it answered. The solution is to use something that is not an analog FXO like PRI or SIP to a carrier. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali Sent: Wednesday, April 27, 2011 7:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Orginate not working well with PSTN lines Thanks for your solution. Anybody can explain me why asterisk is unable to detect ringback tone from PSTN telco ? . Does anybody successed; to make asterisk to detect ring back tone from PSTN telco ? Thanks, Ashik -- _ -- 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] Orginate not working well with PSTN lines
Originate successfully queued only means that the originate action was handed off to asterisk not that is was executed yet. There are other events, depending on which events you are reading, that tell you the call was answered and such. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote: Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890 Response: Success ActionID: ABC45678901234567890 Message: Originate successfully queued -- Remote UNIX connection disconnected Channel DAHDI/1-1 was answered. -- Executing [1234@outbound-ivr:1] SayDigits(DAHDI/1-1, 1234) in new stack -- DAHDI/1-1 Playing 'digits/1.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/2.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/3.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/4.gsm' (language 'en') -- Executing [1234@outbound-ivr:2] Playback(DAHDI/1-1, demo-congrats) in new stack -- DAHDI/1-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [1234@outbound-ivr:3] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' Thanks Regards, Ashik -- _ -- 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 -- _ -- 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
