Displaying 20 results from an estimated 4000 matches similar to: "g729 recording on asterisk using g729 enabledphone"
2005 Jul 07
0
Re: Braodvoice - UK Non Geographic Numbers
asterisk-users-bounces@lists.digium.com wrote:
> http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm
> Of course these are BT retail rates but I fully expect wholesale
> rates based on call prefix will be available for carriers / ITSP
In some countries there's a company (companies?) providing access
to a database which telcos can use to find the rates on this
2005 Jul 28
0
Zaptel rpm spec file with udev support
Hi,
Has anyone written a SPEC file for zaptel, with kernel
2.6 and udev support? I can find some spec files here
and there, but from what I can see they're all kernel
2.4 / non udev...
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 3 3016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 413 65 45
2005 Mar 29
3
Zaptel based timing for VoIP-only Asterisk
Hi,
In a VoIP only environment, Asterisk has to use ztdummy
to have any chance of playing back understandable audio
files (without drops, hickups etc).
I have been using ztdummy to some degree of success, but
I also have a "Wildcard TDM400P REV E/F Board 1" in the
Asterisk machine I'm using. I'm not using this card for
anything at all, but I'm wondering how to set it
2005 Sep 09
4
Huge Echo
asterisk-users-bounces@lists.digium.com wrote:
> In the following setup:
> call coming from a pstn line -> into FXO card -> asterisk -> SIP
> phone
>
> i get an incredible loud echo in the SIP phone (about 0,5-1s)
> (everything i speak into SIP phone microphone i hear in its
> speaker). The person calling from PSTN is not getting any echo.
Make sure you're not
2005 Feb 23
0
logger reload/restart hanging
Hi,
We're running a very old version of Asterisk
(CVS-HEAD-08/03/04) and we're having some
problems with logging.
Our logger.conf has the following:
full => notice,warning,error,debug,verbose
After having started Asterisk, asterisk will hang in
"/usr/sbin/asterisk -rx 'logger reload'" unless some
output has been sent to the file. I can't find
anything on
2005 Mar 15
0
Zombie or soft hangup
Hi,
What does this line of output mean?
Bridge stops because we're zombie or need a soft hangup:
I'm seeing this sometimes... I've looked in channel.c,
but the code is not much more revealing than the
debug line...
--
Andreas Sikkema Rits tele.com
Van Vollenhovenstraat 3 3016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 413 65 45
2005 Mar 07
1
chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
asterisk-users-bounces@lists.digium.com wrote:
> Is there anyone else with the same problem?
Yes, we've seen the same problem. We have found a work
around, but I'm unable to to look into it today.
--
Andreas Sikkema Rits tele.com
Van Vollenhovenstraat 3 3016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 2245540
2005 Sep 27
1
failed make install on Solaris 10
I finally got Solaris to successfully make asterisk, using these
instructions:
http://sunfreeware.com/programlistsparc10.html#gcc33
Now though, when I issue the make install, I get this error:
mkdir -p /var/opt/asterisk/spool/system
mkdir -p /var/opt/asterisk/spool/tmp
mkdir -p /var/opt/asterisk/spool/meetme
install -m 755 asterisk /opt/asterisk/usr/sbin/
install: asterisk was not found
2005 Sep 02
3
DTMF and "breaking through" voice prompts
Has anyone else had problems with users being able to press key tones during
a voice prompt? I have a few users complaining that some systems will not
recognize key presses during them.
using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode.
Thanks
Sherwood McGowan
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2006 Jan 19
1
DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones
> I have seen the following effect in Asterisk, though: where
> it converts
> an inband DTMF (eg coming off a Zap channel) into an
> indication, it mutes
> the audio where that tone is. But sometimes it leaves a
> teeny bit of the
> tone behind.
>
> If you take such a call over say IAX to somewhere and then
> back out a Zap
> channel, you end up with the
2004 Sep 09
2
Fax relaying with T.38
Hi,
We've got endpoints and gateways who have T.38 fax support. We
now use SER and Asterisk to do our routing and other
functionality, but fax doesn't seem to work. Asterisk complains
like this:
Sep 9 09:25:45 WARNING[467828746]: RTP Read too short
Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256)
With lots of the
2004 Sep 17
1
How would you handle a fax without T.38orG.711uLaw?
asterisk-users-bounces@lists.digium.com wrote:
> Isn't it possible to use T.38 for interconnecting hardware gates
> supporting T.38 with asterisk using SIP REINVITE?
> I'm not shure but but think its's might be possible because after
> reinvite traffic goes directly from one gate to anotger, not over
> Asterisk
We've seen a problem here with asterisk. Wehn
2007 Jun 05
1
g729
I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies
Jun 5
2005 Apr 24
2
g729 passthrough?
I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I get a complaint from the server:
-- Call accepted by 66.225.202.72 (format g729)
--
2004 Aug 06
2
Difficulty evaluating the return value of PlayBack (or any other extensions.conf command
Hi,
I just started to "play" with Asterisk today and while I'm
writing some IVR-like functionality in extensions.conf I
would like to take a decision based on whether playing a file
succeeds:
exten => s,2,GotoIf($[Playback(${CALLERIDNUM}_personal) = 0]?3,501)
So if Playback succeeds I want to jump to label 3, otherwise to
label 500. Unfortunately Asterisk doesn't seem
2005 Aug 17
8
DECT gateways
Heya list,
I need some advice/experience.
Some of our customers are asking us about DECT solutions for
their asterisk install. Some others will not go to asterisk
if there won't be a DECT solution.
They now have a Siemens or a Samsung PBX. Those PBX-es come
with a DECT basestation and optionally repeaters etc.
All those basestations speak some own protocol to the PBX,
so we cannot use them
2005 Aug 30
2
How to use * and # as part of numberindialcommand
What is CFU and CFNR?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Michel Koenen
> Sent: Tuesday, August 30, 2005 1:46 AM
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] How to use * and # as part of
> numberindialcommand
>
> > From: "Damon
2005 Sep 22
1
Early Media with Asterisk
Hi :)
I hope someone has a hint concerning Early Media.
The situation:
My Asterisk is connected to small local carrier who works with several SIP
servers.
I traced some SIP headers and find something like this:
Via: SIP/2.0 UDP sip1.provider1.de
In the SDP part I found something like this:
o=- 2268929 0 IN IP4 sip2.provider1.de
c=IN IP4 sip2.provider1.de
If I send
2005 Sep 29
2
Don't call
I have set up extension.conf and sip.con with default
parameter of UNIVOICE server, but Asterisk show this
message when I call a number:
Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899
create_addr: No such host: univoice,Ttr
Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
== Everyone is
2004 Aug 28
4
G729 licenses
Hi, all!!!
What will Asterisk do in the following case:
For example, we have 4 licenses, and have 4
simultaneous calls, using G729.
Will asterisk allow incoming calls from peer,
that can talk G729 and ulaw, and will it
force it somehow to use ulaw in this case?
All phones there in LAN behind Asterisk
prefer GSM codec, so it does transcoding.
So, what I mean is will Asterisk fall back
to use