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Hi,
I have a strange problem, when I try to connect to les.net from our local
asterisk server through Procurve router I seems to be connecting on any port
above 1024 and when I reload sip the port is changing too ...
So I never get 5060? Any ideas on what is going on and how to resolve it?
Hi All,
I defined 3 trunks for a client:
[trunk-100]
...
[trunk-101]
...
[trunk-102]
...
How can i do a trunk hunting ability like this:
[dial-out]
exten = _1.,1,Dial(SIP/${ext...@trunk-100)
if busy try:
exten = _1.,1,Dial(SIP/${ext...@trunk-102)
if busy try:
exten =
On 27 Dec 2008, at 22:54, Razza wrote:
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk
also, I want all the channels to register with asterisk.
I have the FXS channels working fine, I cant acheive that with the
FXO channels, does anyone have any advice or possibly sample
Sriram wrote:
2. Is there any way to block the caller id from appearing on the SIP
Phone ? my trunk is E1 PRI while i used softphones internally - i
dont want my agents to see the caller id - is their any way to block
caller ids from appearing on softphones ?
a) SetCallerPres(restircted)
Please see below Console Messages, Pertinent section of SIP.CONF and
AudioCodes Config.
*Console Messages:*
Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 handle_request_register:
Registration from 'sip:2...@192.168.10.4 sip%3a...@192.168.10.4' failed
for '192.168.10.4'
Dec 28 18:14:45
Hello there!!, Am looking for a manual or documentation that explain
how to buy a DID number and how to configure it with Asterisk, and
when some body call to that DID number Asterisk answer with a
automatic operator
Some body know about this manual? I already search in the web but nothing
Razza,
I have a MP114 FXO/FXS that I have never got to work , even as an FXS,
even though I have several other FXS's that work fine ie Linksys PAP2
etc.. would you put up your config?
PDE
___
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Jerry Geis wrote:
I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri
1.4.7
and I am getting the error:
-- Requested transfer capability: 0x00 - SPEECH
-- Called 23/317506
-- Channel 0/23, span 1 got hangup, cause 99
-- Hungup 'DAHDI/23-1'
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
On Sun, 28 Dec 2008, Abel Monzon wrote:
Hello there!!, Am looking for a manual or documentation that explain
how to buy a DID number and how to configure it with Asterisk, and
when some body call to that DID number Asterisk answer with a
automatic operator
Stating what country you're in
On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote:
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
Matt,
Asterisk version == 1.4.22
dtmfmode == info
calls are bridged through Asterisk (canreinvite=no)
Jonathan
On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell astma...@gmail.com wrote:
On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote:
I am trying to resolve an issue and I believe
I had installed Asterisk 1.4 and when I call to a exist extension, the
voice have noise, but, when I call to a extension does no exist,
asterisk played a voice that say me that extension does no exist, but
without noise
I want I some body can test with a softphone my server,
ip:
On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote:
Matt,
Asterisk version == 1.4.22
dtmfmode == info
calls are bridged through Asterisk (canreinvite=no)
Jonathan
Have you tried setting dtmfmode to 'inband' for both SIP endpoints?
MATT---
On Sun, Dec 28, 2008 at 3:23 PM, Matt
Hangup Cause 99 is:
=
An Information Element or Parameter Does Not Exist or is Not Implemented.
Indicates that the equipment sending this cause has received a message
which includes information element(s)/parameter(s) not recognized
because the
Hi,
Thanks for your prompt reply. Let me clarify some thing on my requirement.
That's not a realistic expectation. How can you presume that because
callerid is xyz that it's always the same person calling? You can
not. Office's routinely have one main number with callerid being the
same for
On Mon, Dec 29, 2008 at 5:51 AM, sasikala kala
sasi_jeyalaks...@yahoo.com wrote:
Hi,
Thanks for your prompt reply. Let me clarify some thing on my requirement.
That's not a realistic expectation. How can you presume that because
callerid is xyz that it's always the same person calling? You
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