:)
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
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
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: "James A. Shigley"
Date: Mon, 28 Dec 2009 12:11:35
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject:
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
>
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
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
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