We worked on this bug today and are expecting to release packages with the fix
soon, possibly tomorrow (Aug 2). The issue arose from a change in features
reload which was back-ported to 1.6.2 and was committed without enough testing
to observe the intermittent crash behavior.
Thanks for your p
On 2/08/2011 4:13 AM, Robert Huddleston wrote:
Thanks -- and did you find a provider with T.38 DIDs? I don't see many
pay as you go providers with T.38
I am not looking for VoIP providers for such functionality and is
subjective to geographic location, however I am of the opinion that one
On 08/01/2011 03:35 PM, Paul Belanger wrote:
On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
You are closing the socket before reading the result of 'Logoff' and
Asterisk is complaining.
Well, he's sending DBPut before reading the result of Login as well.
--
Kevin P. Fleming
Digium, Inc. |
On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
You are closing the socket before reading the result of 'Logoff' and
Asterisk is complaining.
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asteris
Hi guys, I hope you could help me.
I am trying to get connected through AMI but something is not working. Both
php code and manager.conf were working well in asterisk 1.4
1. Sometimes it gets connected and sometimes it doesn't:
== Connect attempt from '192.168.25.241' unable to authenticate
=
Thanks - and did you find a provider with T.38 DIDs? I don't see many pay as
you go providers with T.38
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore
Sent: Monday, August 01, 2011 4:08 PM
To: Asterisk Users Mailing List -
On 2/08/2011 1:02 AM, Robert Huddleston wrote:
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
Thanks
Yes, it works.
I currently have latest firmware installed and it still works in T.38. I
am using UDP trans
Richard,
Thanks for the explanation. You were right about the lack of signalling to
indicate that the call has been rejected, One particular service provider,
instead of signalling rightaway that the call has been rejected, gives a
voice message saying 'The user is busy. Please call later.', and d
My apologies - yes.. Grandstream HT-502...
Apparently finding a t.38 provider is also another struggle...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, August 01, 2011 1:05 PM
T
On 08/01/2011 12:02 PM, Robert Huddleston wrote:
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
You'd be more likely to get relevant responses if you had included the
information about the HT-502 in your message
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On 08/01/2011 07:43 PM, Kevin P. Fleming wrote:
On 08/01/2011 04:12 AM, CB wrote:
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
Are there any plans to include the ISAC codec in Asterisk? Is it
possible or
even desirable? Is ISAC open source (nothing indicates it is from the
WebRTC
web
Richard,
I tried calling the same number outside of Asterisk, by making direct calls
from a landline telephone and a mobile phones. When the user rejected the
call, the call was immediately cancelled.
This implies that for whatever reason, the call reject signal is not
available for asterisk to p
Thanks for feedback. Yeah, tell me about it. Your description is very
accurate of the situation. I can't believe it's in the repo without any
tests done; even the simplest reload. I don't mean to be a whiner but
honestly the repo is a joke with such an obvious flaw for so long
now
On Sun,
> There is no event for Asterisk to recognize. The PROGRESS message just
> says that there is an audio message available for the caller to listen
> to. Asterisk just passes the indication to the peer channel and opens
> the audio path. It is the caller who must recognize any audio message
> that th
Dear Robert
Are you at live IP ???
--- On Sun, 7/31/11, Robert-iPhone wrote:
From: Robert-iPhone
Subject: Re: [asterisk-users] sip attacks
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
Date: Sunday, July 31, 2011
On 08/01/2011 04:12 AM, CB wrote:
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
Are there any plans to include the ISAC codec in Asterisk? Is it
possible or
even desirable? Is ISAC open source (nothing indicates it is from the
WebRTC
website http://www.webrtc.org)?
What do you need i
On 08/01/2011 12:53 AM, Nikhil wrote:
Does anyone know about this...
On 06/20/2011 04:34 PM, Nikhil wrote:
Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables
1. chan->readformat
2. chan->writef
> On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
> > Are there any plans to include the ISAC codec in Asterisk? Is it
> possible or
> > even desirable? Is ISAC open source (nothing indicates it is from the
> WebRTC
> > website http://www.webrtc.org)?
>
> What do you need it for?
>
The possib
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