Displaying 20 results from an estimated 1000 matches similar to: "Re: So you all think VoIP sypply is warm andfuzzy"
2005 Jul 18
9
So you all think VoIP sypply is warm and fuzzy
Here is a letter I sent them for my $150 paper weight.
Dear Voipsupply, As a small service provider, using you company for the
first time, I'm very disappointed that you have removed the
configuration CD that should have been shipped with the Mediatrix 2102
just to get a few more bucks. I have contacted mediatrix and they have
informed me that the CD's is shipped in every 2102. If I
2005 Jul 27
19
Full T38 sip Faxing now Available
Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.
Thanks
Michael D. Schelin
ShellTel
626-814-2354
2005 Feb 26
2
Interface * with ATA from ATA FXS port?
Me again... I have service with a company that does not allow for a BYOD
plan. They will not give out credential or server info either. Is it
possible to run the FXS port of the ATA to an FXO port in *?
The service I have is throug Broadvox Direct using the Mediatrix 2102. I
have tried this using loop start and kewl start. The * box sees the
incoming ring, picks up and starts my dial plan. But
2008 May 05
2
T38 Passthrough Verification
Hi All,
I have 1.4.9.1 setup, with the compiler flags enabled for T38, and
have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes
between devices but can't seem to invoke T38 pt UDPTL. It's enabled
in sip.conf [general] and well as the [peer].
I get an error at the CLI:
WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite
after T38 session not handled yet !
2004 Jul 11
1
mediatrix 1204 hysteria
Hello guys,
I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider
2005 Sep 14
1
Liquidation: Cisco; Polycom; D-Link; MediaTrix, Colubris - Highly Reduced Prices
We have extra equipment that was over-ordered or unused. All of the
equipment is brand new. The equipment has been highly discounted to move
quickly - the last set of equipment sold in 48 hours. If this equipment is
of interest to you, call or e-mail quickly.
Buy on VOXILLA and SAVE $300 each (Cisco routers & switches):
http://store.voxilla.com/customer/home.php?cat=259
For Sale (all new):
2005 Sep 15
2
Fax->Email for Hosted PBX
I'm proposing to install an Asterisk PBX at a collocation facility for a
remote customer. Each of the customer locations will have an SPA-3000
with the FXO port connecting a POTS circuit and the FXS port connecting
a fax machine or red phone.
In addition to voice traffic, the customer has a high volume of incoming
and outgoing faxes.
Would it be possible, using g711 between the SPA-3000
2004 Jun 07
2
Mediatrix 1204 Configuration
I added those lines to my configuration, and i just see with ethereal that my client dial
and the 1204 led turn on and they started to interchange packets, im newbie with asterisk
i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up?
could u send me all the configuration i need step by step?
----- Original Message -----
From: "Wojciech
2003 Sep 02
1
problems with mediatrix 1204 FXO
I'm having a problem getting outbound trunking to work using asterisk
and an external SIP FXO.
7 digit dialing produces the following output:
-- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack
-- Called 5925660@mediatrix-1204
-- SIP/mediatrix-1204-645e answered SIP/mitel-fe17
-- Attempting native bridge of SIP/mitel-fe17 and
2007 Jul 11
1
Access specific port of Mediatrix 1204 from Asterisk
I am attempting to use a Mediatrix 1204 to interface to multizone paging
from Asterisk. I have 4 different paging interfaces and want to connect
each of those 4 to an FXO port on the Mediatrix. The desired result is
to be able to issue some SIP dial string from asterisk, seize the FXO
port on the Mediatrix and then have a speech path.
I am able to place calls over the Mediatrix when it's
2004 Jul 06
2
Mediatrix 1102 Problems
We have a Mediatrix 1102 hooked into the network. Both of the attached
analog phones and all of their features work, but in the CLI we keep
getting "-- Got SIP response 481 "Transaction Does Not Exist" back from
XXX.XXX.XXX.XXX " (Where XXX is the IP address of the Mediatrix ) every
few minutes. I have changed most of the settings in the sip.conf
multiple times and have done
2013 Feb 11
2
[OT] Mediatrix Euro ISDN hangup problem
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi list,
I'm getting a strange problem with a Mediatrix 3631 Gateway connected to
the PSTN via an E1 PRI link configured for Euro ISDN signaling. The
Mediatrix sends incoming calls from the PSTN to an Asterisk server via
SIP: this works fine. But when the caller hangs up, the Mediatrix
doesn't send "Bye" to Asterisk, so the call is
2004 Jul 20
1
Random Dropped Called
I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as
my PSTN interface. I'm experiencing random dropped calls on the
various SIP devices I have tested. Network connectivity to the SIP
devices looks ok, and I have tried a variety of the devices including
all of the following.
Grandstream 286
Grandstresm 486
Sipura SPA 1000
Mediatrix 2102
Some example lines from my logs
2004 Feb 03
1
Mediatrix 1102 Auth
Hi all.
I'm evaluating a mediatrix 2fxs 1102.
seems great (it has also supervised transfer, that's
very needed in office environments and works well).
the only I thing I cannot make work is the auth
to my asterisk server.
If I don't set a password into the mediatrix and
*, I can call out, but still the registration goes wrong.
using a password, nothing works.
I've done some
2005 Oct 07
1
Outbound Mediatrix 1204.
Dear Group,
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my configuration on Asterisk;
exten => _78996.,1,Dial(SIP/${EXTEN:5}@192.168.6.52)
exten =>
2005 Mar 11
1
NuFone Configuration [problem]
Hello,
I am trying to configure the my asterisk box here with the following
**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
***extensions.conf:***
exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN}
I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan.
2004 Sep 03
1
RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
I have the user manual, I'll send it to your email tonight when I'll be in
my home.
I have an APA III-4FXO too, until today I can't put it to work with
asterisk.
Kind regards,
Miguel
Date: Fri, 03 Sep 2004 16:07:59 +1000
From: Jamie Carl <geek@j-code.net>
Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help.
Anyone with user manual?
To:
2006 Jan 13
1
Calls through madiatrix with incorrect disposition
hi guys,
I have an asterisk server and a mediatrix 1204 gateway. I make calls
through the mediatrix unit (only outgoing calls). The problem is, every
call I make through the mediatrix unit is logged in the cdr as
'ANSWERED', even if the call was 'NO ANSWER' in practice.
Any ideas how to make cdr records accurate?
Thanks!
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An HTML
2006 Dec 10
1
Mediatrix 1124 setup
I recently purchased a Mediatrix 1124 from an auction of a company
that went out of business. It came with nothing other than the unit
itself.
In digging thru the Mediatrix web site, and various google searches,
it looks like it only supports SNMP setup, and only with their
software (or the correct MIB). However, Mediatrix doesn't appear to
let you download said software or MIB from
2004 Jan 31
2
Dial via sip gateway?
I'm having a brain fart....
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten => _6X.,1,Dial(SIP/3091@205.22.93.1/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich