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Does this model give you functioning mwi? > -----Original Message----- > From: Ray Jackson [mailto:ray@jacksonz.net] > Sent: Friday, January 05, 2007 3:17 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail personalised greetings using > DB/IMAPbackend? > > > Hi all, > > I am attempting to build a horizontally scalable Asterisk...

All, Is there a way of rejecting a call using SIP in the Asterisk Dialplan? Essentially, I want to look at the called number and if it matches something I don't like I want to send back a SIP response which will not cause the other end to 'hunt'. The response codes that will achieve this are: 401 Unauthorized 403 Forbidden Is there a way of getting Asterisk to send back such a

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

All, Has anybody come across a problem with the MS-GSM codec on Asterisk? We have a client for Windows XP that uses the MS-GSM codec. While the codec seems to work when talking directly to Asterisk, when 2 of these clients talk to each other through Asterisk the sound is all mangled on both ends. I assume MS-GSM is the same as used by Windows Messenger and so my problem may be related to

Hi all, We are looking at VoIP over Wifi and I was wondering if anybody had any ideas around automatically transfering calls after an RTP timeout? The idea is this: a user is on a call with their IP phone and the connection drops (e.g. user walks out of range of their Wifi AP). Using RTP timeout I was hoping rather than just dropping the call I could keep the other party on hold whilst

All, Using SIP with progressinband=yes I get Asterisk to generate the ringing sound for callers. However, I was wondering if it is possible to configure what is 'played back' to the calling party? i.e. instead of just 'ring ring' could I potentially play back a song from an MP3, WAV or GSM file? I'm thinking it would be quite cool to offer a customised 'ring'

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms

We seem to have a problem with Asterisk 1.4 when a client sends through their SDP information but includes encoding parameters on the end of their SDP information. For example some phones send: a=rtpmap:18 G729/8000/1 instead of the usual: a=rtpmap:18 G729/8000 in the SDP... It seems that when the encoding parameter '/1' is included at the end of the rtpmap statement, Asterisk

Hi all, I am attempting to build a horizontally scalable Asterisk deployment and am getting very close to achieving that goal. With Asterisk 1.4 I now have an IMAP backend for Voicemail messages which is great as users can check the same messages either through the voice portal or using Webmail. However, I'm not sure the best way of dealing with personalised greetings such as a