Hi guys,
Leo Ann Boon wrote:
I have a couple of interconnected asterisk boxes connected to several
providers. With one provider in particular (ATP in Australia) there
are two ringing tones heard on outbound calls. It is not the end of
the earth - I am not reselling our services yet - but it is
I'm working from the docs here:
http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
and getting an error doing the ./configure on the iksemel module:
checking for getaddrinfo... yes
./configure: line 20399: syntax error near unexpected token `,'
./configure: line 20399: `AM_PATH_LIB
> you need to add
>
> ;this extension MUST be here for OriginateFailure triggers
> exten => failed,1,Hangup
>
> to your context used for *send too after connect*
Richard,
THANK YOU!! This makes a lot of sense - I don't know why I didn't catch
that before. I can add my SetCDRUserField stuff in
Hi Noah,
Thanks for your reply.
Please clarify one more doubt in extensions.conf file...
is the following dial plan is right way to call another server(frome serverA
to serverB)
exten => _5X,1,Dial(sip/[EMAIL PROTECTED]:6030,15,tr)
exten => _5X,2,Hangup
here for an example the extensi
Lee Jenkins wrote:
After figuring out a problem with AGI and freepascal, I have finished
writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a
small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt.
I'd like to give it to the community (source/binary) and wa
Hi Again Dan -
> Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.
>
> VoiceMaster only authenticates IP and cant have username password based
> authentication which asterisk can do. So i need to take some traffic from
> VoiceMaster to Asterisk and terminate it.
That should
Hi Dan -
Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.
VoiceMaster only authenticates IP and cant have username password based
authentication which asterisk can do. So i need to take some traffic from
VoiceMaster to Asterisk and terminate it.
That shouldn't be a probl
Hi Ram,
Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.
VoiceMaster only authenticates IP and cant have username password based
authentication which asterisk can do. So i need to take some traffic from
VoiceMaster to Asterisk and terminate it.
Let me know
Thanks
Dan
O
Dovid, you're killing me. This after asking if we can't all just be nice to
each other.
On 1/1/07, Dovid B <[EMAIL PROTECTED]> wrote:
Adam and bill are both wrong. The world revolves around me. Geeez cant we
cut the crap (i.e. Happy new year is followed by a response that "hey it
isnt the new
Adam and bill are both wrong. The world revolves around me. Geeez cant we cut
the crap (i.e. Happy new year is followed by a response that "hey it isnt the
new year here yet") If you need the attention find a place where there is a
live TV feed (report) and say "I am a tool, I need attention
On 12/31/06, Adam Jacob Muller <[EMAIL PROTECTED]> wrote:
It's still 2006 here
-Adam
Well, Adam, I guess it is all about you. What does the rest of the world
look like as it revolves around you?
___
--Bandwidth and Colocation provided by Easynews.
Josué Conti wrote:
Always...
Desire that in the New Year that if you really initiate...
It hears the words that always it desired to hear. It pronounces the
phrases that one day it desired to repeat.
It feels the emotion that always waited to feel.
It walks for the tracks that one day it desire
John French wrote:
I'm using Polycom Soundpoint phones and I want to use some extensions beginning with #
for features setup. I'm getting the fast busy "can't match it" signal. I want
to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding
number and turn it o
I'm using Polycom Soundpoint phones and I want to use some extensions beginning
with # for features setup. I'm getting the fast busy "can't match it" signal. I
want to match #50 for call forwarding, for instance, and #505551212 to set the
call forwarding number and turn it on. I have tftp set up
Yuan LIU wrote:
Not sure if anyone experienced the same - or if anyone ever connected
a POTS phone to the "Phone" jack on an X100P card.
The POTS phone rings normally when the FXO receives a call. The POTS
phone can also make outgoing calls when FXO is not holding the line.
This is desired
Hi, I have ip phones Thomson ST2020 and have couple problems with them.So I don't know how to configure voicemail button in phone to get voicemailsfrom Asterisk. In gui configuration window I need to enter URl of voicemail field, butI don't know what is the syntax of this address. Maybe someone c
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