[Asterisk-Users] Outbound IAX calls stop ringing remote phone, yet can still pick up
Greetings, I've recently encountered some strange behavior placing outbound calls using IAX via a VOIP provider. Intermittently, calls placed will ring the called phone a couple times, then the ringing stops. However, even after a few seconds of silence i can pick up the phone and the call is connected. I've tried this with two different VOIP providers (voicepulse and nufone), and have experienced the behavior on both. Any ideas? I would've thought that the voip provider would be responsible for generating the ring tones, but i'm not sure. 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] Re: Outbound IAX calls stop ringing remote phone, yet can still pick up
Greetings, I've recently encountered some strange behavior placing outbound calls using IAX via a VOIP provider. Intermittently, calls placed will ring the called phone a couple times, then the ringing stops. However, even after a few seconds of silence i can pick up the phone and the call is connected. I've tried this with two different VOIP providers (voicepulse and nufone), and have experienced the behavior on both. Any ideas? I would've thought that the voip provider would be responsible for generating the ring tones, but i'm not sure. Regards, Steve add an r to the end to your dial statement - Original Message - From: Stephen David [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 29, 2004 12:06 PM Subject: [Asterisk-Users] Outbound IAX calls stop ringing remote phone,yet can still pick up That will, of course HIDE any BUSY or telco messages. And the caller will never know of they dialed an invalid number or of the number they dialed is busy. Do you think he REALLY wants that? Steve Totaro wrote: I don't know, but sweeping the problem under the rug with r is not a solution except as a last resort. Regardless of if he ends up using r or not he needs to know what other problems r will cause. What we really need is a console output of a call that happens with (hopefully with a note where the ringback actually stops). Thanks, both. However, i was talking about the ringing for the 'CALLED' party, no the 'CALLING' party. I don't care about the ringing for the calling party because its an oubound dialing app using .CALL files. The ringing (periodically) stops after 2 rings or so -- but the CALLEE can still pick up the line and connect the call. also, killing the call progress information for the call is no an option -- i need to know if the lines was busy / no answer. (don't worry, NOT a telemarketing app!) can you think of any reason why the CALLEE's line would stop ringing after 2 rings (sporadically), while the line is still active? ___ 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] Re: call progress - what are the sticking points?
i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) 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? not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. 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] 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] Re: pound key tone generated after call answered?
Message: 1 Date: Sun, 25 Jul 2004 22:15:32 -0400 From: Stephen David [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pound key tone generated after call answered? Reply-To: [EMAIL PROTECTED] Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail (ie. verizon wireless), the call placed by asterisk appears to be immediately generating a tone (upon answer) that the voicemail system is interpreting as a pound key (#). I know this because if i listen to tmp.gsm, i hear the tail end of please enter your password, then press pound., and NOT hello, this is steve, please leave message as i expected. Just in case anyone stumbles upon this in the future, i figured it out. The problem is not that a # key tone was getting generated on answer, the problem was that i was setting the CallerID to the same value that I was dialing. Apparently it is a feature of certain voicemail systems that it detects this and automatically goes to voicemail. that's why if you dial yourself from your cellphone you get voicemail instead of your greeting. also of note is that this only happens when dialing using IAX2 with a call termination provider, and not with ZAP and a POTS line. also note that i've tried two different IAX2 providers with the same resuls. (using zap and pots lines is not preferrable in this case) one more note is that i've disabled the 'beep' before the recording starts (commented out lines in app_record.c and re-built code), just in case that was causing it -- no change in behavior, though. any thoughts on what could be causing this, or how to further troubleshoot? Regards, Steve whew... 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] pound key tone generated after call answered?
Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail (ie. verizon wireless), the call placed by asterisk appears to be immediately generating a tone (upon answer) that the voicemail system is interpreting as a pound key (#). I know this because if i listen to tmp.gsm, i hear the tail end of please enter your password, then press pound., and NOT hello, this is steve, please leave message as i expected. also of note is that this only happens when dialing using IAX2 with a call termination provider, and not with ZAP and a POTS line. also note that i've tried two different IAX2 providers with the same resuls. (using zap and pots lines is not preferrable in this case) one more note is that i've disabled the 'beep' before the recording starts (commented out lines in app_record.c and re-built code), just in case that was causing it -- no change in behavior, though. any thoughts on what could be causing this, or how to further troubleshoot? 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
