Your soundcard doesn't have much to do with MOH. In fact, you can run MOH
entirely without a soundcard. Therefore, OSS vs. ALSA won't affect your MOH
performance.
You might want to make sure that your musiconhold.conf file is correct, and
that you really do have MPG123 and NOT MPG321. res_music
> "Dan" == Dan <[EMAIL PROTECTED]> writes:
>> Agreed. Jeremy McNamara of Nufone.net is the top dog in Asterisk
>> VOIP and long distance.
Dan> Hi, How can you subscribe to this service? There is no web page
Dan> available to do it.
I emailed them at [EMAIL PROTECTED], as per one of the pa
You didn't mention the distro you are using. I'm wondering if you are
using one of the distros that leans towards the alsa drivers. If so,
then chan_oss would have problems.
On Sun, 2003-07-20 at 19:16, Stuart Hirst wrote:
> I am having a problem getting music on hold working one of my servers.
>
Stuart Hirst wrote:
When I put a call on hold the CLI shows moh starting but nothing is
played. No errors are reported whilst starting moh.
I have been trying lots of different things for hours now without success.
Anyone got any pointers ?
What does your musiconhold.conf file look like
Hello!
Well, so much for mailing me off-list: not a single person did! In other
words, you've already seen the results of my request:
The options are:
Nufone.net
Cost: 2.9 cents/min for both outgoing long distance and incoming 800
calls. Service is pre-paid.
Advantages: Extremel
Title: Message
I am having a
problem getting music on hold working one of my servers. I have had this working
on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV
1.5ghz.
I have installed
mpg123 which seems to be working fine but when I start *, I get the following
er
Hi,
I'm having a problem with DTMF tones from my SIP client apparently crashing
the chan_capi driver. However I'm not sure whether this is a bug or
misconfiguration on my part: if I set "softdtmf=1" in
/etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support
DTMF detection?
The
This is generally indicates a problem with the licensing process (which
is severely flawed and full of bugs) on your server... Did you make it
through the registration process OK?
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of An
Is it stable enought?
I mean around 30-40 Incoming SIP connections.
Or i must trash the cisco and put Asteriisk/Digium/speex box?
Jeremy McNamara wrote:
> You have to run a console with the G.729 due to the voice age library
> lameness. We run safe_asterisk with a TTY and it seems to be fine.
>
Would this work with SIP / H323 phones??
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 21, 2003 1:11 AM
Subject: Re: [Asterisk-Users] Self parked but avaliable
> On Sun, 2003-07-20 at 06:43, [EMAIL PROTECTED] wrote:
> > Is
On Sun, 20 Jul 2003, Linus Surguy wrote:
> Hi all,
>
> We're currently running a PSTN -> SIP gateway with Asterisk. We also run
> IAX/SIP -> PSTN.
>
> We have performed a test where the call is routed
>
> UK PSTN -> Digium E1 card -> Asterisk GW -> SIP G.711 -> FWD -> X-Ten
> softphone
>
> T
Send email to
[EMAIL PROTECTED]
I did this late Friday afternoon and Jeremy had me set up in
very short order. (I gave him wrong contact info Friday but he
IM'd me the needed account info Saturday morning.)
I've got problems with my own IP connection but aside from
that the service just works.
Maybe the agent stuff?
Mark
On Sun, 20 Jul 2003 [EMAIL PROTECTED] wrote:
> Is there any way I can define an extension that I can call which will park my
> call so that I can listen to hold music over the speaker phone, but then if a
> real call comes in for me, it prompts me to press a key to ac
Yo,
I'm trying to get Asterisk working with Messenger 4.7. After skimming
through the list-archives, I've got it to register to my Asterisk-box
and can make calls. Unfortunately, there's no audio from the Messenger-
side of the call to the other caller. I can hear the caller in Messenger,
thoug
Hi all,
We're currently running a PSTN -> SIP gateway with Asterisk. We also run
IAX/SIP -> PSTN.
We have performed a test where the call is routed
UK PSTN -> Digium E1 card -> Asterisk GW -> SIP G.711 -> FWD -> X-Ten
softphone
There is no echo at the softphone end, but severe echo on the PSTN
You have to run a console with the G.729 due to the voice age library
lameness. We run safe_asterisk with a TTY and it seems to be fine.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Try launching asterisk like this:
screen -d -m asterisk -vvvcn
Aparently there is some bug in the codec.
- Just
Try launching asterisk like this:
screen -d -m asterisk -vvvcn
Aparently there is some bug in the codec.
- Justin
On Sun, 20 Jul 2003, Anton Tinchev wrote:
> Before few days i bought few g.729 licenses.
> When i try to load the codec, asterisk crahses.
> I tried with and without oh323 module,
hi all
sorry, this price was wrong
www.global-gateway.net does 2.55c/min for end users and down to 1.921
for wholesale
roy
On Sun, 2003-07-20 at 17:46, Roy Sigurd Karlsbakk wrote:
> We're using http://www.global-gateway.net/
>
> I've compared their pricing with nufone.com, and for what I can se
We're using http://www.global-gateway.net/
I've compared their pricing with nufone.com, and for what I can see,
they're quite a bit below (.us @ 21c/min). They do not, however, have
.us number termination. Their website sucks but the voip works. we have
an IAX2 trunk over to their .uk site.
roy
You have something drasticly wrong somewhere... 64 is SLINEAR and
chan_h323 does nothing with SLINEAR frames.
1 is G.723.1
4 is G.711 u-law
check your config...ensure your allow'ing the proper codec's.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Having problems to connect another device using c
On Sun, 2003-07-20 at 06:43, [EMAIL PROTECTED] wrote:
> Is there any way I can define an extension that I can call which will park my
> call so that I can listen to hold music over the speaker phone, but then if a
> real call comes in for me, it prompts me to press a key to accept the call, and
Is there any way I can define an extension that I can call which will park my
call so that I can listen to hold music over the speaker phone, but then if a
real call comes in for me, it prompts me to press a key to accept the call, and
if I do then it takes me out of parking can connects me to t
Having problems to connect another device using chan_h323.
When G723.1 or G711: log says:
NOTICE[15376]: File channel.c, Line 1325 (ast_set_read_format): Unable to
find a path from 1 to 64
NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to
find a path from 4 to 1
WARNING[
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