Displaying 20 results from an estimated 90 matches similar to: "SIP, DTMF, and AudioCodes Mediant 2k"
2003 Nov 23
2
SIP Express Router & Asterisk
Greetings...
We've been having some interoperability issues between Asterisk and an
AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000
somewhere. So, I've been pondering using iptel.org's SIP server (SIP
Express Router) as a "front end" for PSTN calls going out to the Mediant,
while using Asterisk for everything else.
Has anyone done something similar, or
2003 Sep 28
0
TE410P timing and multiple, different spans
Greetings...
I have a TE410P with four T1's going into it. Things look roughly like
this:
#1 Goes to PBX -- we're responsible for timing
#2 E&M span to telco 1
#3 PRI span to telco 1
#4 PRI span to telco 2
If I set primary sync source to span 2, users report strange echo,
distortion, and crosstalk problems, which sound remarkably like frame
slippage on spans 3 and 4. If I set
2003 Oct 21
1
"Defragmenting" mailboxes
Does anyone have a quick and dirty script for defragmenting mailboxes?
i.e.:
-rwx------ 1 root root 80553 Oct 20 16:27 msg0000.gsm
-rw-r--r-- 1 root root 218 Oct 20 16:27 msg0000.txt
-rwx------ 1 root root 781164 Oct 20 16:27 msg0000.wav
-rwx------ 1 root root 79360 Oct 20 16:27 msg0000.WAV
-rwx------ 1 root root 7260 Oct
2003 Nov 05
1
A real-life production scenario
Since it's all the craze, I might as well post our current Asterisk
usage. :-)
EQUIPMENT:
- Beefyish box (dual Xeon 2.4GHz, gig of RAM, more-than-adequate disk
space, etc) in a 1U chassis.
- A second, slightly less beefyish box of specs I don't have handy right
now, also in a 1U.
- 2xTE410P
CONNECTIONS:
- 1 PRI to telco for local outbound/direct-dial inbound, 300 numbers
2003 Oct 07
3
Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says:
astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r
I have tried to release it with soft hangup Zap/1
& also soft hangup
2003 Nov 05
12
Mediatrix 1204
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0
2003 Jun 22
2
How can I log SIP debug messages to a file?
Hi everybody,
I want to read to debug messages and try to interpret them but they happen
too fast, how can I log these guys to a file, or is there a file like this
already?
I checked the /var/log/asterisk but there isn't much interesting there yet?
How can i turn on logging for SIP,IAX and other things?
Thanks,
Umut
2003 Jul 11
1
SIP call from one extention to another
Hi
I am trying to call from Linphone on extention 109 to Xlite on extention 108
and I get this error
----------------------
to 216.75.167.18:5068
WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application
'Dial ' for extension (sip, 108, 1)
== Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43'
---------------------
Can you tell me what
2003 Jul 23
2
SIP info
I was wondering what are the values for sending dmtf
via sip info.
I mean, when I use dtmf relay via sip info, the sip/sdp message
contains a Signal=X where X is the dmtf.
That's ok for dtmf 0-9 . but what when dtmf is * or # ?
we must send signal=# ?
I ask that because I noticed that budgetones phone sends out
* as signal=10 and # as signal=11 . but asterisk
don't detect them, 'cause
2003 Oct 16
1
VoIP Monitor
Hi all!
I am looking for some free software to monitoring all the calls that are being
done in my network. Which telephone are connected, how long are the calls,
quality of service, bandwidht,etc.
If someone knows about a good one, plesea tell me.
Regards,
Mireia
2003 Oct 24
1
Questions about Zapateller and Privacy Manager
Hey all...I'm just getting my * setup and right now all I have is an FXO
but no FXS. I wan't to get rid of telemarketers by having * pick up the
phone if there is no CID present, give the caller the Zapateller tones
and then ask the user to input their phone number via Privacy Manager
(yes I realize that this won't get us any where given that I can't
re-ring the phones without FXS
2003 Jul 23
1
Cisco 7960 upgrade from SKINNY load
Here's a clip of comments lifted from a Cisco bug list. This will
be perhaps useful to those of you who have just purchased a Cisco
phone off eBay.
JT
-------------
(1) Short problem description:
Documentation on how to load SIP image on phone with skinny software
(2) Longer problem description (what happens):
If the phone is loaded with the Cisco Skinny code, then there is a
small
2003 Oct 17
4
Using channel banks
Hello Everyone,
What kind of hardware setup would I need to do if I want a T1 connection to PSTN
and have 48 users in office with analog phones. Will something work if I have a
T410P card in asterisk and have one T1 going to PSTN and other two to a channel
bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks.
Deepak
2003 Jul 24
4
the 'pound' and '#' are the same?
Hi,
I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message:
%vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail
2003 Oct 21
9
Free g.729.1 implementation
1st. - I'm from Poland, we don't have (yet, and hopefully forever) software
patents.
Is there any free g.729.1 implementation for asterisk? I want to use it for my
private use (dialing into inet->PSTN gateway), and I don't want (now) to buy
codec, as I don't know if I will be using this service in future (now I just
want to test it). Any solutions? Maybe even
2009 May 15
0
Mediant 1000 audiocodes and Trixbox
Hi,
This is my first experience with a mediant 1000 and an Asterisk Trixbox.
the mediant has 12 FXOs and 12 FXSs, and I want to use it them all.
I will have extensions connected to the FXS ports, and lines to the FXO.
Can anyone guide me, please?
regards,
--
Guillermo Garron
"Linux IS user friendly... It's just selective about who its friends are."
(Using Ubuntu, Debian,
2007 Jun 21
2
mediant 2000 with asterik configuration
Dear all
anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it
---------------------------------
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2007 Jun 20
0
asterisk with mediant 2000 trunk
Dear All
I want to integrate asterisk with mediant so anybody have configuration for this setup
[asterisk]----------[mediant]------[avaya]
this is my setup so what is the basic configuration for this setup
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An
2010 Apr 10
1
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Hello my friends,
I want to make fax work in the following scenario:
My versions are:
Asterisk 1.4.21.2
WANPIPE Release: 3.4.7
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
The E1 pri is connected to our Sangoma A102DE, we also have a SIP
Mediant Audiocodes 1000 where we have some fax machines connected to
fxs ports, what we need is to make fax machines through mediant
2004 Sep 16
2
Audiocodes Mediant 2000
Hi FOlks,
I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to
work with Asterisk through SIP or H323.
But since I don't the product manual, it's being a little hard.
Anybody would the manual in PDF(file or URL) to indicate to me?
Thanks a lot,
Isamar