I have a server that is receiving a disconnect during recording of long
incoming messages. The connection is via a SIP gateway and when the gateway
sees no RTP for 5 mins, it hangs up the call. I enabled
transmit_silence_during_record but I see no RTP being sent from Asterisk to
the gateway durin
parameter,
a file name. But the attempts I have tried seem unsucessful. I have tried
dialing out and then calling SendFAX and calling SendFAX before the dial.
No success.
Can someone please provide me with an extensions.conf example of how to use
SendFAX?
Thank you.
Jonathan Augenstine
parameter,
a file name. But the attempts I have tried seem unsucessful. I have tried
dialing out and then calling SendFAX and calling SendFAX before the dial.
No success.
Can someone please provide me with an extensions.conf example of how to use
SendFAX?
Thank you.
Jonathan Augenstine
Matt,
Asterisk version == 1.4.22
dtmfmode == info
calls are bridged through Asterisk (canreinvite=no)
Jonathan
On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell wrote:
> On 12/28/08, jonathan augenstine wrote:
> > I am trying to resolve an issue and I believe it is my configuratio
I am trying to resolve an issue and I believe it is my configuration. The
scenario is that I have a SIP detected on the server. The dial plan then
makes a local connection to another part of the dial plan. The new dial
plan extension then places another SIP call out to a SIP phone. When the
cal
Have you checked out OpenSBC (www.voip-info.org/wiki/view/*OpenSBC)?*
On Fri, Dec 12, 2008 at 6:19 PM, Steve Edwards wrote:
> One of the above is frequently used to front-end Asterisk.
>
> I used OpenSER to front-end a farm of Asterisk servers and was very happy
> with it. The ability to take a b
FYI >> I was informed by A. Minnesale that app_confcall was originally
developed for Asterisk 1.2. He stated that there would probably be a
significant amount of work to update it to Asterisk 1.6.
Jonathan
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I am trying to build app_confcall and it is failing. Are there known build
issues with this module. I am running Asterisk 1.6.0-beta9.
Jonathan
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To UN
Is it possible to create extensions in the voicemail.conf remotely by using
the manager interface. I cannot seem to find any documents or examples
describing that capability.
Jonathan
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Have you verified that ztdummy is loaded?
On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote:
> Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no
> hardware interfaces installed gives me this error. Im a bit new to
> this so any help will be appreciated.
>
> == Parsing '/et
I have had very reliable inbound/outbound service from Junction Networks
(www.junctionnetworks.com). The one time I did have an issue, it was
resolved quickly. During my testing I concluded that BroadVoice (my
partner refers to them as NoVoice) was unreliable (approximately 40% of
all of our test
l Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
> Augenstine
> Sent: March 17, 2006 5:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Question about meetme app
>
> A locked conferen
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
> I have a quick question about the MeetMe app. A locked conference means
> what exactly?
>
> A) That people can't join anymore
> B) That everyone is muted
Try this:
musiconhold.conf:
[stream2]
mode=mp3
directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr
extensions.conf:
exten => 1234,1,Answer
exten => 1234,2,MusicOnHold(stream2)
exten => 1234,3,Hangup
On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote:
> Ok, I'm tearing my hair out
Barix Instreamer takes RCA in and MP3 or ulaw stream out. Asterisk can
use either for MOH.
On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote:
> Been around asterisk for two-plus years, but need a little input from the
> list on this topic.
>
> Have a potential client that wants to replace th
You can try Voxboned(www.voxbone.com) if you need inbound only.
On Tue, 2006-01-24 at 09:13 +, scott wrote:
> Hi
>
> Does anyone know a UK Voip Proivder that will give me more than 1 telephone
> number and point it to my sip account.
>
> www.SipGate.co.uk are great but they only allow 1 te
Here is where you will find the answer to all of your questions:
http://www.asterisk.org/
http://www.voip-info.org/wiki-Asterisk
Jonathan
On Sat, 2005-12-24 at 01:34 +0500, Faheem Ahmed wrote:
> I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I
> need to know initial configur
Cause No. 31 - Network disconnect (Normal, unspecified)/Special
intercept announcement: Call blocked because of group restricitons
It looks like a telco configuration issue. Your provider probably has
toll free block on your trunk(s). You should be able to call them and
ask to have it enabled.
Has anyone successfully connected a Digium T100P to a Zhone Z-Plex 10 24
S/O? I have been unsuccessful in getting the T1 to sync up. I have
searched the documentation and concluded that a cross-over cable and
ESF/B8ZF configuration on both hardware should have cleared alarms but that
does not
Can anyone tell me if they have successfully deployed the X100P in India or
any where in Southeast Asia?
Thank you,
Jonathan
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check out:
http://www.voip-info.org/wiki-Asterisk+phones
At 12:23 AM 10/15/2004 -0700, you wrote:
Hi,
After reading up on the Asterisk, I have a
question:
1. Is there a software phone running on PC as a client that
is compatible with Asterisk?
My reason for asking is that I wonder if I can
Another solution would be to keep the discussions on topic and open up a
separate mailing list for people interested in open discussions.
Jonathan
At 07:17 PM 9/26/2004 +0100, you wrote:
I was somewhat concerned reading Mark's posting earlier today.
Obviously, things are very bad in the US at the
Not the cheapest ($75-80) but they look interesting.
http://ipphone.eezeephone.com/
Jonathan
At 03:10 PM 9/26/2004 -0300, you wrote:
On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman <[EMAIL PROTECTED]> wrote:
> Hi guys,
>
> I know this isn't strictly about Asterisk, but it is related...
>
> I am lo
Asterisk on an SMP system. I cannot find that info now and most of the
reading seems to indicate that SMP stability is good. Does anyone know of
the warning I read?
Thank you.
Jonathan Augenstine
[EMAIL PROTECTED]
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