Displaying 20 results from an estimated 1300 matches similar to: "mediant 2000 with asterik configuration"
2007 Jun 26
1
call fail from audiocode to sip trunk
Dear ALL
I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000
[auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1
When i call from audiocode MP -124 phone i got this error
-- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")
2015 Sep 25
2
Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)
Does anyone have any information for me?
Welinghton.
Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>:
> Hello!
> ?
> I am setting up an Asterisk server with a Mediant 1000 (Audiocodes)
> to make external links. Does anyone have any manual or instructions on
> how to proceed?
> ?
> Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?
2007 Jun 22
2
asterisk 0 dial outgoing call
Dear all
i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is
[sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN]
now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give
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
2007 Jan 15
2
Audiocodes Mediant 1000, Polycom, and no ringback on transfer
I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any
2007 Jun 25
2
Rining 180 and 183
Dear all
I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya
[asterisk]-----[mediant 2000]--------[Avaya]
when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of
2010 Apr 11
0
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode
Thanks James,
What i need is to make the fax machines connected to the audiocodes mediant
1000 be able to send and receive fax throught Asterisk (connected to a pri)
I know it's not reliable, but it should work at leaste, what should i do on
Asterisk and Mediant to make this work?
Im quite confuse with all these fax issues :S
Thanks in advance
>
> Message: 11
> Date: Fri, 9 Apr
2007 Jul 04
1
call transfer not working
Dear all
I have install asterisk 1.2.x and it is working fine my setup is like
[*]-------[Mediant2k]------------[Avaya]
Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task
2008 Jan 29
1
smaba + ldap + privilages
Dear all
I have smb+ ldap setup not everything is fine but i want to assign some right to perticuler Group so they can change TCP/IP properties and change system time and do some other right
Is it possible to give some privilages to normal users ???
$ cat ~/satish/url.txt
http://www.linuxbug.org
2007 Sep 06
2
FAX machine connect with audiocode SIP device
Dear all
I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device
---------------------------------
Ready
2008 Feb 14
1
multiple login deny
Dear all
I want to deny multiple login in domain with single user account what is the solution for it ??
$ cat ~/satish/url.txt
http://www.linuxbug.org
_____________________________________________________________________________________________________
---------------------------------
Get the freedom to save as
2013 Jul 24
1
Mysql Support int Asterik-11
Hi,
I was having question about mysql driver support ( not odbc).
Do we still need the asterisk-add-on to be installed for mysql support.? If yes, Which version should be used and from where I should get it?
Thanks in adavance.
----
Thanks & Regards,
PrashantAbhang
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2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and H.323
Hi List;
All we know that in voice, there are a type of
communications between endpoints, for example: in some
communications we do a proxy for media and signaling
while other communications we do a proxy for only
signaling.
Where I can determine these things in Asterisk if I am
using SIP and if I am using H.323?
Regards
--------------
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
2007 May 10
1
call transfer to asterik.. asterisk as an end point
Hello All.
I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience.
I want to use asterisk for call park/pickup and have configured openser
to relay calls made to ruri 700-720 to asterisk running on
localhost:5069
Call flow:
phone A calls phone B (both phones are polycom)
Phone B answers
then phone b
2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and
Dear Alex;
Thanks for your kindly reply.
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
Another thing, do u mean that it is easier (better) if
we need H.323 endpoint to talk with SIP endpoint then
we use full
2005 Mar 04
0
Asterisk with mediant 2000 - facing problems
Hi,
I have been using/working on asterisk for some time now and presently
was trying to configure asterisk to work with digium cards. It worked fine
with the fxo/fxs cards, but now i'm trying to get it working by interfacing it
with mediant t1 port. no avail .......
anyone out there got it working, what particular configuration used
on mediant (isdn signalling, framing, coding etc ??)
2006 Oct 14
0
SIP trunk from an Audiocodes mediant 1000
Hi,
I am configuring an audiocodes Medant1000 to talk to my asterisk box.
So far I have successfull in landing a single call from mediant to my
*box. my sip conf is as follows:
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[3911700]
type=friend
host=dynamic
dtmfmode=info
secret=blah
context=sip
where 3911700 is my E1 telephone no. in my extensions.conf I have
exten =>
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 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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