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Hello all: Can anyone help me with finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/6b26b235/attachment.htm

I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk "outgoing" directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...n experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...IP phones (small setup, say 4 phones) at the same time. 1) I would like to pickup up any phone and the ringing should stop (of course) 2) Put on hold and pick up on a different phone set. Do I need special phone features to achieve this? E.g. would the Grandstream 100 do it? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...s the file with extension XHTML to be passed to the phone and not return a HTTP 406 error (File type not supported by your browser). Hope is helps others. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Phil Menico Sent: Friday, August 25, 2006 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS Thanks, but we have reasons to want to make it work with IIS. Anyone have a hint of what is the issue? -----Or...

...I develop an external application (say a Call Center application or a GUI management application) that uses Asterisk data is that also GPLd? I understand if I add code to Asterisk, but what about external interfaces? Where is the seperation here of the Cathedral and the Bazaar? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...cables to the phone and >PC which kills a major cost saving for going VoIP.. > >Later.. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...es to be the "outside" caller and go to IVR? What would the outside phone's "dial out" plan do. I assume the configuration files affected would be extensions.conf and sip.conf. If someone has an example of a couple of lines of .conf would be appreciated. Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...g to Bridge". I wait a while but doesnt work. There is no communication between the outside phone and the one on the S100U. There is no sound card in the machine. I assume its not the problem. The machine is an RH9 with 2.2Ghz and 256RAM. Any ideas or help would be appreciated. -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...t comes to speed. It may just be gnome or kde but it is slow launching apps. Does anyone know if Asterisk will compile under other distros. Many people are recommending http://www.gentoo.org/ and I am considering it. What is it about RH9 thats different when compiling Asterisk? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...video package. But I thought to ask here first if someone has seen this before. [root@localhost asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot open shared object file: No such file or directory [root@localhost asterisk]# which ps /bin/ps Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

...aded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a document to setup as well as the API calls (and a book). Anything like this for Asterisk? Thanks for your patience. -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Is there any kind of Configuration guide available? I've been trying to get a SIP Soft phone to work and all I get is: NOTICE[81926]: File chan_sip.c Line 4716 handle_request): Registration from '<sip:xxx@xxx.xxx.xxx.xxx> failed for 'yyy.yyy.yyy.yyy' I know I'm missing something in the configuration. But don't know much about * yet. Thanks, Kyle --------------

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

...1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Based on dialing plan the system should be able to route the call to whatever channel supports dialing that number. Thank you. Phil Menico