On 07/28/2012 05:43 PM, Mike wrote:
what are folks using for PRI gateways these days? Obviously there's lots
of folks using TE410s and related cards, which work well, and I know
reasonably well.
However, anyone using anything standalone that stands out as being
particularly stellar?
Anything
I'm getting the following warning with SOME calls on 1.8.14.1 Is it a cause
for concern? Is there a way to fix it? I can't tell for sure if it is
impacting calls or not.
WARNING[27796]: chan_sip.c:20497 handle_response_invite: just did sched_add
waitid(4077) for sip_reinvite_retry for
On 28-07-2012 18:23, Matthew Jordan wrote:
You stated that you need to add some custom headers, which you can do with the
MESSAGE_DATA function. In order to activate the forwarding, you could
do something like this:
exten = send_msg,1,NoOp()
same =
Hello Tiago,
if you read spanish, maybe this post can help you.
http://www.voztovoice.org/?q=node/549
Regards
Bakko
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- Original Message -
From: Tiago Vasconcelos tiago.o.vasconce...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Sunday, July 29, 2012 4:19:00 PM
Subject: Re: [asterisk-users] How to send a SIP MESSAGE outside a call
Matthew, you used the name send_msg which I found confusing
- Original Message -
From: Mike Diehl mdi...@dominion.diehlnet.com
To: asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Sunday, July 29, 2012 3:39:05 PM
Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
On Saturday 28 July 2012
27.07.2012 19:25, Matthew Jordan пишет:
Hi Dmitry!
Hello!
So, our original conversation is here:
http://lists.digium.com/pipermail/asterisk-video/2012-April/003621.html
As I said in our previous conversation, we don't currently have plans
to implement a re-transmission of a new source's
