Cory Forsyth
2006-Jun-28 10:35 UTC
[Asterisk-Users] asterisk -> my cell phone's voicemail sound problems
When I fail to pick up a call from Asterisk to the PSTN to my cell phone and let it go to voicemail, the sound quality is always really bad. When I call my cell phone's voicemail a few minutes later, it's really garbledy and sounds clipped or something. I've tried using Monitor to record the sounds that are being played to my cell's voicemail, and the monitored sound sounds fine when I open it up on my Mac using Quicktime and listen to it. It also sounds fine if I answer the call and listen to it live. Any idea what could be the problem? I'm using BackgroundDetect to figure out when the voicemail prompt finishes, but other than that nothing fancy is going on. thanks, Cory
Hi, I used a FXO-Gateway to connect my VoIP to PSTN, using inband mode for DTMF. It can work properly if I use this dialplan: exten=> 100.,1,Dial(SIP/xxx.xxx.xxx.xxx/111) 111 is a line number in the gateway. but, I can't get the PSTN number that the caller dialed from the gateway. So I tried to send the DTMF from asterisk when dialing using this dialplan: exten=> _1.,1,SIPdtmfMode(inband) exten=> _1.,2,Dial(SIP/xxx.xxx.xxx.xxx/111,,D(${EXTEN:1})) for some reason, the dialplan above can't work. My assumption is that the DTMF received by Gateway is different from the one that asterisk sent. I'm using g711-alaw codec as stated in my sip.conf: [general] disallow=all allow=alaw ... I already tried using the relaxdtmf but it didn't seem to work. So now I tried to capture the DTMF that the caller made while dialing but I can't find a way to do it. I'm using asterisk 1.2.9.1. Any solution or a pointer to the problem is welcomed. Thanks in advance, Armand