[Asterisk-Users] Call progress from sip gsm gateway to pri interface - doesn't get through
Hi, we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP- GSM Gateway Call comes from PBX through Parlay to Asterisk and it routes it over SIP to GSM gateway. GSM gateway gives back call progress (it takes some time to ring or get through), but this info won't get back to Parlay on ISDN PRI interface (Digium PRI card), so Parlay after some timeout disconnects call We guess that this setup should work, but we're not sure. Anyone with working setup like this? Anyone with experience of call progress getting from SIP to PRI or BRI interfaces ? Any advice or pointer to more info ? Thanks in advance, regards, Rob. ___ --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] Call Progress Analysis
Hi All, I am using Asterisk 1.0.7 with an X101P analog card which is connected to an Alcatel pbx. My problem is that when I place outbound calls on the zap channel, Asterisk returns a connect event as soon as the phone starts ringing. This means that Asterisk is not being able to do Call Progress analysis on the zap channels. I have tried setting 'callprogress=yes' in zapata.conf butit made no difference. This problem is not there with SIP and IAX channels. Here's my zapata.conf: [trunkgroups]; define any trunk groups [channels]; hardware channels; defaultcontext=defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesrelaxdtmf=yesbusydetect=yesbusycount=6callprogress=yesprogzone=uk group=1callgroup=1pickupgroup=1immediate=no ; define channelssignalling=fxs_kschannel = 1 ___ --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
[Asterisk-Users] Call Progress Analysis
Nitin Joshi wrote: Hi All, I am using Asterisk 1.0.7 with an X101P analog card which is connected to an Alcatel pbx. My problem is that when I place outbound calls on the zap channel, Asterisk returns a connect event as soon as the phone start ringing. This means that Asterisk is not being able to do Call Progress analysis on the zap channels. I have tried setting 'callprogress=yes' in zapata.conf but it made no difference. This problem is not there with SIP and IAX channels. I have the same problem with Digium TDM cards. I've been doing pretty extensive research and found no solution. Look at my mail [Asterisk-Users] [Fwd: call status with FXO], few mails ahead of yours. ___ --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
[Asterisk-Users] Call Progress Detection
Title: Call Progress Detection We have done some answering detecection coding that can differentiate answering machines and live answers. We're having problems with operator intercepts. Asterisk is showing them as No Answers, Does anyone have any suggestions on how to properly differentiate between Operator Intercepts and No Answers in the call progress detection? Toby [EMAIL PROTECTED] 949.278.1896 ___ 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
[Asterisk-Users] Call Progress Analysis
Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and executes the command there as if nobody answered the call. I need this because I want to have a follow-me application that dials different phone numbers or extensions based on the call status. Thank you in advance for your help. Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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] call progress - what are the sticking points?
I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? Thanks much! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call progress - what are the sticking points?
Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? Thanks much! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call progress detection
Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its possible using telephony boards (ie. Dialogic/Intel), but don't know about *. I have experimented with callprogress=yes in zapata.conf, but not sure if that was intended to cover what i describe above. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] call progress detection
On Sun, 2004-07-18 at 20:38, Stephen David wrote: Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its possible using telephony boards (ie. Dialogic/Intel), but don't know about *. I have experimented with callprogress=yes in zapata.conf, but not sure if that was intended to cover what i describe above. callprogress is to detect pickup, ringing, hangup, and busy signal on analog lines that don't support a complex enough signalling to support a computer on the other side. What you need is something like a ecording looking for silence post answer. AGI supports record with silence detection. Once you detect the specified amount of silence, you can play your message. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call progress on x100p
downloaded and compiled today's CVS (04/08/2004) tried using callprogress on Via mini-itx (running RedHat Linux 9) if callprogress is set to yes on x100p, an i call the line connected to x100p, asterisk would execute the first app and will wait forever. anyone had success using callprogress? thanks. __ Do you Yahoo!? Yahoo! Small Business $15K Web Design Giveaway http://promotions.yahoo.com/design_giveaway/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Progress
Hi Mark, I have implemented a procedure for automatically calls from the client-side (IaxClient - E100P) What I want to do is to detect the call status from the client-side. Meaning, if the line is busy/unavailable/fax log the status and proceed to next call. Is this possible with Manager API ? Any solution/idea ? Best Regards, Marin __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call progress when making call using ATA via iconnecthere
Just curious as to whether or not there's anyone else out there who is using iconnecthere behind a completely-NATted asterisk system. My system, using code from approx a week ago, works just fine with iconnect, except that I get no call progress information (if that's the correct terminology). I dial, wait a bit, and if the person on the other end picks up I just suddenly hear them, without hearing any ring tones first. I fake this with r in the context, but I would much rather have it be meaningful, since this way I get ringing tones even when iconnect has sent one of its infernal occasional 480 messages. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
