search for: getdtmf

...amp;m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now in progress -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/8-1' Nov 23 19:08:59 WARNING[1828889]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now in progress -- Hungup 'Zap/8-1' -- Starting simple switch on ...

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other than those two problems the voice q...

...difference.. It seems like Asterisk never gets any digits from the upstream switch. I don't think the upstream switch gets a wink from Asterisk, but I am not sure. Here's what the console log shows. -- Starting simple switch on 'Zap/1-1' File chan_zap.c, Line 3772 (ss_thread): getdtmf on channel 1: Operation now in progress == Unknown extension 's' in context 'default' requested -- Playing 'ss-noservice' -- Hungup 'Zap/1-1' Incidentally, inbound calls on the PRI are immediately disconnected when inbound caller-id info is present. The E&M trun...

...e line, and goes to extension s, no digits passed. If I make it dial my phone directly when it tries to outdial I hear a quick blip of tone, then silence, then it disconnects. I tried switching over to a em instead of em_w signaling in zapata.conf and I get a warning in asterisk: "ss_thread: getdtmf on channel 24:success" when it tries to dial. I'm going to build up a test system with a spare TE110P I have spare to play with this tomorrow instead of using my live system, with the newest libraries, but maybe this is a signaling issue I'm just not understanding, something this card...

...:15 WARNING[25806] chan_sip.c: No such host: 5040 Oct 24 09:13:15 WARNING[25806] channel.c: No channel type registered for '' Oct 24 09:13:15 NOTICE[25806] app_dial.c: Unable to create channel of type '' (cause 66 - Channel not implemented) Oct 24 09:18:31 WARNING[25905] chan_zap.c: getdtmf on channel 39: Operation now in progress Oct 24 09:19:44 WARNING[20740] chan_sip.c: Unknown SDP media type in offer: image 5006 udptl t38 Oct 24 09:22:17 WARNING[20711] channel.c: Avoided initial deadlock for '0x83b3788', 10 retries! Oct 24 09:25:04 WARNING[26095] chan_sip.c: No such host:...

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another