Hi,
I have watched a phenomen, that I can not explain... maybe one of you
can see the reason why the call failed, and if the cause
is the Snom Hardphone, or the asterisk, or the SIP-Provider... the debug
log given below is all I have...
What does "Setting SIP_ALREADYGONE on dialog.." mean?
th
Ok, it seems like I don't have g729 codec intsalled, can I install this
codec within asterisk 1.2 or just 1.4 and 1.6 are supported?
On Fri, Nov 12, 2010 at 2:56 PM, khalid touati wrote:
> Hi Guys,
> I have a the following issue when I ma trying to place a call through my
> voip provider, I am u
Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)
Nov 12 14:31:
Sorry this is a list for the Asterisk GUI Project. I think you may
have better luck on the FreePBX list / forums.
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Le
Hi,
On freepbx (GUI), whatever reason number fails we always get 'all circuits
are busy' audio.
Does anybody know how to get far end audio when we dial wrong number or when
it's busy or unallocated number or failed with some other reason.
Thanks,
Deepika
--
__
Hi,
Please disable firewall and SElinux.
2010/7/9 Philipp von Klitzing
> > SEND >> 0.0.0.100:5060
>
> ?!
>
>
> --
> _
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> New to Asterisk? Join us for a live
> SEND >> 0.0.0.100:5060
?!
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asteris
Hello:
Here is my sip and extentions configuration and the log of x-lite, because i
don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i
hope you can help me.
SIP.conf
[default]
include=>anexos
include=>anexos1
include=>anexos2
[anexos]
exten=> 100,1,Dial(SIP/100,0)
exten=
Robert La Ferla wrote:
> After so many rings when the party does not answer, my SIP phone says
> Call Failed. Why doesn't it just keep ringing?
>
> Here's the dial plan rule:
>
> exten => _NX,1,Dial(SIP/[EMAIL PROTECTED],,r)
> exten => _NX,n,Hangup()
>
Not that it's the caus
After so many rings when the party does not answer, my SIP phone says
Call Failed. Why doesn't it just keep ringing?
Here's the dial plan rule:
exten => _NX,1,Dial(SIP/[EMAIL PROTECTED],,r)
exten => _NX,n,Hangup()
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B.Srinivasa Rao wrote:
Hi,
Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a
Hi,
Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which succe
Hi!
I frequently get errors like "Call failed to go through, reason X" in
/var/log/asterisk/messages
Are the reasons explained anywhere? I did not find any info.
Thanks,
Carlos
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Hi!
I frequently get errors like "Call failed to go through, reason 0" in
/var/log/asterisk/messages
Are the reasons (0,3 and 5 in my case) explained anywhere? I did not
find any info in the wiki.
Thanks,
Christian
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