On Sun, Oct 11, 2009 at 8:03 AM, James Stocks wrote:
> OK. For anyone finding this thread, the problem exists in Asterisk
> 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem.
Sorry, I lost your last response in my inbox... Your phone configs
look fine. The only thing that
On 3 Oct 2009, at 20:38, James Stocks wrote:
> On 3 Oct 2009, at 16:37, Jonathan Thurman wrote:
>
>> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks
>> wrote:
>>> Hi everyone,
>>>
>>> I hope someone can help me with a problem I'm having with Cisco 7940
>>> phones on the SIP 8.12 image. When I plac
On 3 Oct 2009, at 16:37, Jonathan Thurman wrote:
> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks
> wrote:
>> Hi everyone,
>>
>> I hope someone can help me with a problem I'm having with Cisco 7940
>> phones on the SIP 8.12 image. When I place a call from one of the
>> handsets, the call proceed
On Sat, Oct 3, 2009 at 6:17 AM, James Stocks wrote:
> Hi everyone,
>
> I hope someone can help me with a problem I'm having with Cisco 7940
> phones on the SIP 8.12 image. When I place a call from one of the
> handsets, the call proceeds as normal for 20 seconds and is then
> terminated by Asteri
Hi everyone,
I hope someone can help me with a problem I'm having with Cisco 7940
phones on the SIP 8.12 image. When I place a call from one of the
handsets, the call proceeds as normal for 20 seconds and is then
terminated by Asterisk (1.4.26.2):
[Oct 3 10:08:55] WARNING[1650]: chan_sip
Where do I get oej's patch, and how do I install it?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Brodmann
Sent: Tuesday, June 05, 2007 2:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls being dropped
I just solved a similar problem on my asterisk box. i just enabled nat=yes
and removed the externip from the nat portion in sip.conf. Try it.
On 6/4/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest fi
We have a similar problem at our place, since a few months.
oej, mentioned a patch he has made after the release of asterisk-1.4.4. So
we're
all desperately waiting for asterisk-1.4.5 to be released; unless you want
to install
from svn.
2007/6/4, Compnet Bobby <[EMAIL PROTECTED]>:
We have
that becasue the reinvite is using a private ip probably..
sip debug
pastebin the results..
look in the re-invite part..
On 6/4/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3 gr
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest
stable firmware) and are having a few problems. We have a basic menu that
transfers calls to different extensions. The problems can be found on all
exte
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