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I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is

...hat if a) it streams constantly and b) my math is correct. Thanks, Doug > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Douglas Garstang > Sent: Wednesday, February 22, 2006 4:18 PM > To: jaugenstine@stoneeyes.com; Asterisk Users Mailing List - Non- > Commercial Discussion > Subject: RE: [Asterisk-Users] Streaming Music On Hold > > Thanks. I got it working. Yay. > > Now, it seems that Asterisk is very fussy with the streams. A lot don't > work, especially when the URL...

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

Is it possible to create extensions in the voicemail.conf remotely by using the manager interface. I cannot seem to find any documents or examples describing that capability. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081015/82b359c2/attachment.htm

I am trying to build app_confcall and it is failing. Are there known build issues with this module. I am running Asterisk 1.6.0-beta9. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081016/d1eda77f/attachment.htm