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...a long lived agi it may > not (although it can take more ram). > > > > > -- > Trixter http://www.0xdecafbad.com Bret McDanel > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > http://www.sacaug.org/ Sacramento Asterisk Users Group >

...e different channel drivers that could potentially use that codec, whatever the method it would have to be generic ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/200...

this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

hi all, i m using debian to run my asterisk gateway.I want to make some customization in voicemail application.For that i need to modify voicmail's .C(source file) file. can any body tell me where exactly all .C files resides in the system.......... thanks tejas __________________________________________ Yahoo! DSL ? Something to write home about. Just $16.99/mo. or less.

HI: what is command on console to know how many calls are sending in the same time? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC

If i get a 8XX number, my provider told me that they will send all the calls he gets. But due to bandwidth and asterisk capacitiy in this particular installation, the system is only able to handle 27 incoming calls. How in my dialplan do I regulate, sending a busy signal, when my system hits 27 incoming calls? thanks, -- ------------------------------------------- Erick Perez

I want to create a VoIP solution to allow many members of a closed community to talk to each other (one on one) via soft phones. In many ways, what I want is not unlike Skype, except that it would allow for relative anonymity and be open only to a select group. The system should support as many as 500 simultaneous users located on different continents. Is Asterisk an appropriate tool to use in

...en the box is completly idle sip does not experience this problem, it is only when there are a few concurrent calls. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/200...

I have searched and found a couple examples on how to put the app_cepstral.c into Asterisk but it isn't working. I obviously am not understanding the examples that I have found. "Copy the app_cepstral.c file to your asterisk source tree (apps folder) . You'll also need to add a lines like these to the Makefile in that directory: APPS+=app_cepstral.so" I added it into

...call quantities this large, and other factors. Anyone that has done this before I would appreciate hearing from you. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/200...

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep being restarted. I wrote this shell script to fool Asterisk into numbering my voicemails sequentially

Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Sun 1/22/2006 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion

Hi All I have some grandstream phones registered to my asterisk and all internal, external, voicemail services etc are working very well. I am not sure that it is a problem more so an annoyance. If someone dials my extension number or external DDI while I am already in a call rather than skipping to the next priority in the dial plan for example voicemail the line continues to ring and while in

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

Hi All I am trying to simplify a dialplan for a few thousand users. Would what I have below work? If someone dials exten 710001 would it go through answer and then to the macro to try dialing the SIP phone thats registered on 710001 and then onto voicemail if no answer or not signed on? exten => 71XXXX,1,Answer() exten => 71XXXX,2,Macro(71macro,${EXTEN}) exten => 71XXXX,3,Hangup()

I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps