Thanks, I'll try and see if it solves my problem
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Hi,
I've tried a few times wiki instructions for setting up a streaming MoH, by
creating stream.mp3 file in ../moh/stream folder. It never worked. I need to
setup streaming MoH and please if someone has a working example, please help
me to set it up.
--
Zeeshan A Zakaria
On Sun, Nov 26, 2006 at 04:30:39AM -0500, Zeeshan Zakaria wrote:
Hi,
I've tried a few times wiki instructions for setting up a streaming MoH, by
creating stream.mp3 file in ../moh/stream folder. It never worked. I need to
setup streaming MoH and please if someone has a working example,
On Fri, 24 Nov 2006 17:34:52 +0100, Alban wrote:
I had some troubles once with db9 and RJ45... It was due to an
impedance pb. As I remember, you have to change the impedance in
the adaptor (with adding some R-C)... I haven't done it, and my
line was half-working (sometimes yes, sometimes
It' seems to be RTP problem.
sendto() in rtp.c fails to send rtp packets.
When I change channel from H323 to SIP, no problem.
Any idea?
Regards,
Jason.
-- from /var/log/asterisk/full--
[Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission
error of packet 39407 to
Any ideas or tutorial on creating your own DIDs without buying them bulk from a
Telco. I have the Asterisk server being hosted in a data center in California.
I guess I can order PRI through them but how can get DID from other states onto
their system.
-- Original message
On 24 Nov 2006, at 22:40, Lachek Butalek wrote:
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
Hi - is support for FastAGI built in by default or do I need to
configure anything within Asterisk to make it understand how to call
FAGI scripts in the dialplan that contain agi://localhost/myFAGI.agi?
Thanks,
Bret
--
Bret Schuhmacher
[EMAIL PROTECTED]
678.528.2385
President/CTO, Last Mile
Hello everybody,
I'd like to ask, is it possible to get extension's hint status from dialplay?
Something like GetHint(10)?
I've only found show hints CLI command...
Thanks a lot in advance!
with best regards
Nik
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Bret Schuhmacher a écrit :
Hi - is support for FastAGI built in by default or do I need to
configure anything within Asterisk to make it understand how to call
FAGI scripts in the dialplan that contain agi://localhost/myFAGI.agi?
Works pretty much out of the box :-)
Vincent,
I do something similar to what you're doing. However, I use the CID number
as the astdb family, allowing me to assign multiple attributes as the keys.
It requires some maintenance, so I also wrote a php script for the
management. You can find it here:
quicktime player does it without adding any codecs.
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
By inverting the relationship, I found it easier to focus on the source of
the call and the treatments I want to apply. I can also wipe out entries
by family name and remove all attributes in one operation using database
deltree.
where is the calling button?
I do not have it. I just can call typing numbers. The manual says that I
should press # at the end of the number, but it does not work
- Original Message -
From: Brett Crapser [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am looking for a toll-free US 1800
DID which can be setup quickly . I have seen nufone there quality is
very good but they charge for 30 seconds minimum ( others do 6/6 i
guess
) . east coast gateway
server preffered . . Plz lemme know if you have some suggestions i
want it to be setup very
Dear
I installed asterisk 1.2.12.1 from local network every thing is doing
fine.but when I connect to it trough internet on port 5060 it rings but no
rtp .when changing the port every function works .
What should I do ,in need to have port 5060 listening
Regards
I have been real happy with voxbone.
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, November 25, 2006 4:35 PM
Subject: [asterisk-users] DID Provider
I am using DIDx.net as my DID provider but they don't seem to get their
Hmmm. If you have access to the users list then dont you also have access to
the biz list ? Digium set up the lists this way for a reason ;)
- Original Message -
From: Mark Phillips [EMAIL PROTECTED]
To: Anthony Rodgers [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List -
Please post the requests on the asterisk biz list.
- Original Message -
From: Vicky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, November 26, 2006 8:35 PM
Subject: [asterisk-users] Looking for toll-free US did
I am looking for a toll-free US
I like a challenge. I'll let you know if I come up with anything.
On 11/26/06, Vincent Delporte [EMAIL PROTECTED] wrote:
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
By inverting the relationship, I found it easier to focus on the source
of
the call and the treatments I want
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
I like a challenge. I'll let you know if I come up with anything.
My eternal gratitude if you find something :-) And don't forget to update
the VoIP wiki so others can benefit too.
___
At 12:00 25/11/2006 -0700, Anselm Martin Hoffmeister
[EMAIL PROTECTED] wrote:
Do try
asterisk -rx database put cidname 12345676 \Me - cellular\
or
asterisk -rx 'database put cidname 3871263 Me - home'
These quotations seem to work.
Yup, I should have tried before posting. Thanks.
At 12:00 25/11/2006 -0700, Michelle Dupuis [EMAIL PROTECTED] wrote:
Try using smartCID (www.generationd.com). You'll get the benefit of
ranges of numbers mapping to single ID's (good for corporate blocks),
action field for blocking/accepting calls, etc).
Neat, although 411.com won't do as
I have done simple ael2 script, tak doing lookup in asterisk database like:
find full numer, if cidname isn't found, substract one digit from right
and try again, and so on
PJ
Vincent Delporte wrote:
Anyhow, at this point, I could successfully import all the name +
number records, and
On Sunday 26 November 2006 12:04 pm, Androtech wrote:
where is the calling button?
I do not have it. I just can call typing numbers. The manual says that I
should press # at the end of the number, but it does not work
On Saturday 25 November 2006 09:38 am, Androtech wrote:
Hi all,
I
We've found spandsp to be quite reliable and robust in recent months,
particularly 0.0.3 versions. We have had issues with particular fax
machines though.
Has anyone else had experiences where some fax machines get errors
sending to spandsp and spandsp get's Unexpected DCN sending to some fax
Hi All,
Wrote up a little howto this weekend on getting Asterisk 1.4 and the
GUI working on my stock Gentoo box from source.
http://www.voipphreak.ca/archives/351-Asterisk-1.4-GUI,-MYSQL-CDR,-Gentoo-Howto.html
Matt Gibson
___
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Hi Guys,
Wrote up a small script for rolodex'ing with Asterisk this weekend.
It's really nothing fancy, but it's useful, and it's fast. Requires
apache/php on the Asterisk server for now.
http://www.voipphreak.ca/archives/337-Asterisk-1.4-PHP-Rolodex-Howto-Script.html
Matt Gibson
Interesting.. I'll call them and confuse them tomorrow :P They always
love when I call haha. At any rate.. clearly Asterisk though the
channel was already in use.. but it wasn't.. so with CPE Yields set,
they will send the call along even if Asterisk says the channel is
busy? What will
Hi,
I have an IVR that sounds just fine and dandy over ZAP. However, when
I dial in through an 800 number from a provider that I connect to via
IAX I get this 'blip' in the sound file. At first I thought it was
just packet loss, but it happens at the exact same spot every single
time. There
I just posted my MWI routines, subject to GNU, GPL. They will set or reset
the MWI light based on an extension number. Also you can cron a job to fix
all MWI lights based on the current voicemail status for all extensions with
the final script I enclosed.
http://www.siliconvp.us/index.php
:-)
I'm trying to use Asterisk (v1.2.11) make a callback that dials both
legs of the call into a Meetme() room together, but I keep getting
conf-invalid messages.
I created a callfile (/var/spool/asterisk/outgoing/out.call) that
specifies a Local channel (extension) which contains a
Hello ppl,
Is it possible to send a REDIRECT from an Asterisk box, to an incoming
call??
e.g. A calling B, via Asterisk,
Asterisk sends redirect to A to contact C.
cheerz
- Ben.
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DO any one succeeded making strp works with asterisk
Please tell me how
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set to
10 sec.
[voicepulseincoming]
exten=_X.,1,Answer
exte=_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
I had the same problem and found I needed a (for you example)
exten = s,n,waitexten
after the last background. This is shown in many examples and in others it
is not. Very confusing but I think adding this will work for you.
Doug
On Mon, 27 Nov 2006, Jeronimo Romero wrote:
I'm experiencing a
Hi
I've got a few 8 port Junghanns BRI ISDN cards. Dialling in and out is
working fine but the Telco's busy or invalid number indications are not
being passed through to the user. I have priindication=passthrough in my
zapata.conf but this doesn't seem to help. I'm using Asterisk 1.2.13,
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