Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio in this direction ------>-------------------->--------------------> [test server that plays .wav file then DTMF digits] -----PSTN-------[Asterisk]-----SIP----[Cisco 7960] ----<------------------------------------call setup in this direction ---------<---------------< ---------------< The Asterisk is set for DTMF=inband , codec g711ulaw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030812/79698d97/attachment.htm
they might be too lound regards Martin On Tue, 12 Aug 2003, Lee Goodman wrote:> Can anyone explain why this is happening? > > I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted > > ----->------------->--------------------audio in this direction ------>-------------------->--------------------> > [test server that plays .wav file then DTMF digits] -----PSTN-------[Asterisk]-----SIP----[Cisco 7960] > ----<------------------------------------call setup in this direction ---------<---------------< ---------------< > > The Asterisk is set for DTMF=inband , codec g711ulaw
Ok, I think I figured out the problem If both the phone and asterisk are using RFC2833, I get this DTMF problem. But, If I set both the phone and Asterisk to DTMF=inband, the DTMF tones sound much better. (I verified the DTMF status of the call by doing "sip show channel ?" in the CLI) Is there a known problem with RFC2833 in Asterisk? (Note, this would only happen when Asterisk acts as a SIP endpoint in the Call, like when a SIP phone calls out through the FXO port not when the call is just passing through the server). There is also a thread about how the XTEN softphone seems to have a DTMF problem. XTEN says they fixed it, but people (including myself) still see the problem. I wonder if there is a bug in Asterisk support of RFC2833? One of the XTEN users says that one way the problem shows up is if you send a string of DTMF's that are the same number (1001 fails , but 1234 works). I have seen the same issue on a Cisco phone ascessing VM on an Asterisk server. Lee Goodman ----- Original Message ----- From: Lee Goodman To: asterisk-users@lists.digium.com Sent: Tuesday, August 12, 2003 4:32 PM Subject: [Asterisk-Users] Weird DTMF issue Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio in this direction ------>-------------------->--------------------> [test server that plays .wav file then DTMF digits] -----PSTN-------[Asterisk]-----SIP----[Cisco 7960] ----<------------------------------------call setup in this direction ---------<---------------< ---------------< The Asterisk is set for DTMF=inband , codec g711ulaw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030813/d4350bcc/attachment.htm
Ok, this one has me stumped. This setup was working fine Friday and now today it's just stopped working. Details: Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s). Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and out of this box to the PSTN is via IAX2. At some point between Friday and today DTMF stopped working right. Specifically, when you call our main # and are at the IVR, only the first digit you dial is recognized. For example if I try to dial "81" this is all I get for debugging: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMF Subclass: 8 Timestamp: 02123ms SCall: 00020 DCall: 00009 [1.2.3.4:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02123ms SCall: 00009 DCall: 00020 [1.2.3.4:4569] As soon as the "8" is received the Background app stops playing as you'd expect, but it stops recognizing any more digits, and eventually times out and errors out with an invalid extension '8'. Even worse, if you try to dial anybody's direct extensions (2xx) now you end up in the support queue after it times out since the queue is option "2" (yeah I know that's a stupid IVR design, but I had to mimic the old PBX I didn't set up.) I've tried this through two different call paths, one through the PSTN and one direct from my house asterisk system (SIP/IAX2 end-to-end). It behaves the same both ways. The strange part is, while the "invalid extension" message is being played by Playback() all the digits I hit *are* recognized, as they show up in the iax2 debug output. It's only in the Background() app that this seems to be a problem. Any suggestions would be greatly appreciated. This has our IVR totally busted and I've tried everything I can think of so far. -- Joshua M Thompson <funaho@jurai.org>