:)
On 31 October 2011 15:36, salaheddine elharit
wrote:
> thank you so much all works without issue now
>
>
>
> 2011/10/31 Christian Gansberger
>>
>> Hello,
>>
>> You have to disable RTP-Encryption on your Snom under Identity, RTP.
>> It is set to on per default.
>>
>>
>> On 31 October 2011 13:3
thank you so much all works without issue now
2011/10/31 Christian Gansberger
> Hello,
>
> You have to disable RTP-Encryption on your Snom under Identity, RTP.
> It is set to on per default.
>
>
> On 31 October 2011 13:33, salaheddine elharit
> wrote:
> > hello list
> >
> > i have installed
Hello,
You have to disable RTP-Encryption on your Snom under Identity, RTP.
It is set to on per default.
On 31 October 2011 13:33, salaheddine elharit
wrote:
> hello list
>
> i have installed asterisk 1.8.7.1 and i have configured 2 account for sip in
> order to do internal call
>
> when i use
hello list
i have installed asterisk 1.8.7.1 and i have configured 2 account for sip
in order to do internal call
when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from
223 to 222
but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to
snom320 but the issue i c
Behalf Of Juan E.
> Rodríguez
> Sent: Monday, December 28, 2009 12:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP Issue
>
> Is ddwhome defined in global context?? If so, then you should use global
> function.
>
2009 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Issue
Is ddwhome defined in global context?? If so, then you should use global
function.
Paste asterisk log to check.
Saludos,
Juan E. Rodríguez
-Original Message-
From: &qu
on
Subject: [asterisk-users] SIP Issue
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On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:
> Alright I have a SIP phone located off premises with a very annoying
> issue.
>
>
>
> Well I say a sip phone it is actually two phones hooked to a Cisco Spa
> 2102
>
> Link: http://www.cisco.com/en/US/products/ps10026/index.html
>
Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Each phone being a different line/extension.
Alright either line can ALW
Jason,
What type of phones are you using? I originally started getting this
error when I got the Cisco 7961Gs (prior to dumping them and going with
all Polycoms). It turned out to be some setting in the XML provisioning
boot file (although I can't remember which one). Once I went to a
minima
Nat?
On 8/6/07, Jason Walker <[EMAIL PROTECTED]> wrote:
>
> I am getting this error
> [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum
> retries exceeded on transmission [EMAIL PROTECTED] for seqno
> 102 (Critical Response)
> [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944
I am getting this error
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for seqno
102 (Critical Response)
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging
up call [EMAIL PROTECTED] - no reply to our c
Michael,
Where in your extension definition to you dial a channel (SIP, Zap, or other)? You
are missing the dial entry.
-sb
-Original Message-
From: Lists [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 10:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip Issue
Hi all I am having some issues with a gs 100 phone. It is on the same
network as my * server. There is no firewall.
In extentions.conf
exten => 5,1,Answer
exten => 5,2,MusicOnHold(default)
When I dial 5 from the sip phone
-- Executing Answer("SIP/mlh-2e75", "") in new stack
-- Executing
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