Re: [Asterisk-Users] DTMF and ivr systems
Hi, I too am experiencing the same problem you have. I am using inband DTMF processing with ULaw (G711) and like you I notice that Asterisk seams to be passively listening to the line waiting to hear a DTMF. When it hears a DTMF it mutes the handset and regenerates my original DTMF (in a very short burst) which often gets missed by the remote party. This is especially true for IVR systems. I haven't come across a configuration option that keeps Asterisk from muting the handset and regenerating the original DTMF. Perhaps if Asterisk saw that the active channel was using inband processing with G711 it could leave everything alone and just let the user's dtmf go through unfettered! If this isn't possible for some technical reason I wonder if there is a configuration option that allows the user to set the duration of the regenerated dtmf? Maybe a developer will see this and can comment. Best regards, Monty On Thu, 29 Jun 2006, Shane wrote: Hello, Ther's probably a simple answer to this but I've searched around and haven't located anything as yet. Is there a way to have DTMF tones passed through Asterisk without it messing with them? I am using a tdm21b card and when I call an ivr system from the telephone handset (routed over sip or iax2) such as telebanking, the ivr has trouble recognizing tones. When I tested this with a remote party, I was told tones were breaking up. For example, a long press would result in a click, some silence and a small dtmf on the remote end. Triggering a speed dial didn't go well either as he heard only a few tones. I have dtmfmode=inband in sip.conf and have tried relaxdtmf=yes in zapata.conf. I realize Asterisk does need to detect dtmf for things like call parking but can it just pass the audio to the other side with no regard for whether it's dtmf digits? IE. long press results in long tone, etc. Best, Shane ___ --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] Help with using Asterisk with PlusNet in the UK
Hello, I hope someone who has been successful in getting Plus.Net's VOIP service to interface with Asterisk might be able to help. For some reason I can't seam to register or make outgoing calls. If anyone would mind posting their "register" line as well as the Plus.Net context in the sip.conf file that would probably help me figure out what I need to put into my sip.conf. I've seen references in this lists's archives saying that at least a couple of people have it working but they didn't say how. Thanks in advance, Monty ___ --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] DTMF recognition inconsistent in Asterisk
Hello, I am experiencing a strange problem and I am wondering if anyone may have some pointers as to how to overcome it. I have an account with VoipTalk here in the UK which I have connected to my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected to my Asterisk box using both methods. The problem is when I dial into my Asterisk box via my VoipTalk incoming PSTN phone number from a landline or mobile Asterisk does not consistently recognise DTMF's. I have my VoipTalk context for incoming calls set to dial an extension in my house then forward on no-answer to Asterisk voicemail. The idea is to: --go into the voicemail box --go to administrative options --place an outgoing call --dial a local extension that activates the DISA function --Once in DISA, enter a DISA password and then it's possible to dial any other locally defined extension. When I hear my voicemail greeting I press star to get to the prompt where I can enter my voicemail password. Asterisk seams to consistently recognise both the star and my password. So far, so good! Then, I press "4" for administrative options and "3" for place an outgoing call. Again this works fine. The problem however is when I attempt to dial the extension assigned to my DISA function. More often than not, Asterisk seams to miss my DTMF's. In the event I do get to my DISA password prompt, it more often than not misses DTMF's there as well. I have this same problem no matter if I use IAX or SIP to connect to VoipTalk. When connected via SIP I have the DTMF signalling set to "inband" which seams to work as I am able to get into my voicemail box as I said above. Can anyone offer a suggestion as to why it seams like Asterisk is inconsistent with recognising DTMF's? Does it use different recognition routines in different function s of the programme? Is there a way to heighten the recognition sensitivity? Any ideas would be very much appreciated! Thanks in advance Monty ___ --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 application in Asterisk V1.07
Hello, Thanks for the message. The exact version numbers and conf info follow: My Asterisk: /usr/sbin/asterisk -V = Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k Debian/AMD64 package file name/version: pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2_amd64.deb Extension.conf: exten => 8800,1,Meetme(8801|aciMps) meetme.conf: [rooms] conf => 8801 Friends Asterisk: /usr/sbin/asterisk -V = Asterisk-CVS-head Note: Above version seams strange but that's what it returned. It was downloaded via CVS on June 27th 2005. extension.conf exten => 2003,1,Meetme(2003|aciMps) meetme.conf: [rooms] conf => 2003 Does that help at all? Thanks, Monty On Tue, 28 Jun 2005, Moises Silva wrote: i think that the thing that really matters here is wich version of Asterisk are you using exactly. I dont know wich version the latest debian package is using, and i dont know wich version from CVS your friend has compiled. Also, its needed to show the extensions.conf configuration of your friend. Specially the options passed to meetme best regards On 6/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear: "After the tone, say your name and then press the pound key." We both have used virtually the same Meetme configurations.(The FWD 514 extension works the same way) I believe I have all the necessary sounds but am really quite stuck here. Please help! I am using the latest Debian/AMD64 package and my friend compiled from cvs source I believe. My settings follow: extension.conf: exten => 8800,1,Meetme(8801|aciMps) ; ext-8800 accesses conf-room 8801 meetme.conf [rooms] conf => 8801 Thanks for any help, Monty ___ 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 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] MeetMe application in Asterisk V1.07
Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear: "After the tone, say your name and then press the pound key." We both have used virtually the same Meetme configurations.(The FWD 514 extension works the same way) I believe I have all the necessary sounds but am really quite stuck here. Please help! I am using the latest Debian/AMD64 package and my friend compiled from cvs source I believe. My settings follow: extension.conf: exten => 8800,1,Meetme(8801|aciMps) ; ext-8800 accesses conf-room 8801 meetme.conf [rooms] conf => 8801 Thanks for any help, Monty ___ 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] Newbee, help with cdr/odbc/mysql logging problem
Hello, I have now been playing with Asterisk for about a week and absolutely love it! Unfortunately I seam to be having a problem with cdr records and mysql via the ODBC method It would appear to all be set up properly, and I am able to successfully. log inbound/outbound activity into my mysql cdr database. All the inserted values appear to be correct and meaningful but for the Duration and Billsec values. They seam to almost always receive a value of 2147483647. This is very strange since I also still have the cdr-csv module logging to /var/log/asterisk/cdr-csv/Master and it gets correct/meaningful Duration and Billsec values. I've tried all kinds of things to see if I could figure out what I have wrong - including stopping the cdr-csv logging incase two logging facilities can't work at the same time but no luck! Has anyone else run into this, or does anyone have any guesses? I'm stumped! Thanks in advance, Best regards, Monty ___ 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
