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...m to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090204/24e09e33/attachment.htm

Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,

...the queue app, or am I setting this up wrong? I did try adding penalties for the members (i.e. no penalty for SIP/100, a penalty of 1 for SIP/112, 2 for SIP/102, etc.) That just resulted in only SIP/100 being rung. So, what am I missing/doing wrong here? :) Jeremy -- Jeremy Gault, KD4NED <jgault@winworld.cc> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: jgault@winworld.cc

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The

...along with the user-specified number. However, on the IP phone, it simply shows "unknown" Is this just us, or is PrivacyManager not working correctly anymore? Any help would be appreciated. Jeremy - -- - ----------------------------------------------------------------- Jeremy Gault, KD4NED | Network Administrator/Telecomms +1 (423) 303-2562 voice | WinWorld Corporation +1 (423) 472-9265 fax | http://www.winworld.com/ - ----------------------------------------------------------------- -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFD8JzIXRhQkRPmcc8RAn9k...

...XXXXX dialing and actually using a Dial to our upstream carrier, still with the U option, and same results. So, I honestly don't think it is an issue with using Local channels. Any thoughts? Am I just blatantly missing something? Or is something broke? Thanks in advance. :) -- Jeremy Gault, KD4NED Senior Engineer, Voiceopia office: (423) 509-8000 x 102 toll-free: (866) 928-4078 x 102 fax: (423) 244-0565 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140717/49be4b00/attachment.html>

...sponsored by Easynews.com -- > >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Jeremy Gault, KD4NED <jgault@winworld.cc> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! _______________________________________________ --Bandwidth and Colocation sponsore...

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956