search for: alijawad

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2008 Nov 20
1
Voicemail in Real Time
...do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [999alijawad at a2billing:1] VoiceMail("SIP/alijawad-08aaf0f0", "999alijawad at a2billing|u") in new stack [Nov 20 12:17:08] WARNING[22277]: app_voicemail.c:2862 leave_voicemail: No entry in voicemail config file for '999alijawad' -- Executing [999alijawad at a2billing:2] Hangup(...
2008 Apr 22
3
Parsing incoming extension till first @
Hi All When I dial a number it reaches the asterisk switch as abc at xyz@123.com what I need to do is to parse the abc and send it to my pstn gateway as in exten => _.,2,Dial(SIP/${EXTEN}@pstn.gw) this does work but I do have a varying number of numbers before the @ exten => _.,1,Dial(SIP/${EXTEN:0:12}@pstn.gw) Well can I use some kind of regular expression to take all numbers before