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 ports)
At the Hotel they dial there local extension lets say 1234 then the 1204 directs
them to our Asterisk which then sends the call to the working IVR. I need to get
this working with the least amount of hardware expense!
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 ports)
At the Hotel they dial there local extension lets say 1234 then the 1204 directs
them to our Asterisk which then sends the call to the working IVR. I need to get
this working with the least amount of hardware expense!
On Wed, 5 Nov 2003 14:25:12 -0500, Ariel Batista <abatista@avionica.com> wrote:> 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 ports) > > At the Hotel they dial there local extension lets say 1234 then the 1204 > directs them to our Asterisk which then sends the call to the working > IVR. I need to get this working with the least amount of hardware > expense!I have used the Mediatrix 1204 to terminate a POTS line. It does work OK. I've had some problems with caller ID not showing up all the time, but otherwise it's been pretty solid. The configuration, however, was perhaps the most horrible VoIP-related task I've ever done. -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net
---------- Original Message ---------------------------------- From: Ryan Tucker <rtucker@netacc.net>>> 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 >I have used the Mediatrix 1204 to terminate a POTS line. It does work >OK. I've had some problems with caller ID not showing up all the time, >but otherwise it's been pretty solid. > >The configuration, however, was perhaps the most horrible VoIP-related >task I've ever done. -rtWould have a sample script that you used! I have been trying to get it working now for 2 days. It is very hard to get working!> >-- >Ryan Tucker >Network Engineer >NetAccess, Inc. >1159 Pittsford-Victor Road >Bldg. 5, Suite 140 >Pittsford, New York 14534 >585-419-8200 >www.netacc.net >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >
> > Would have a sample script that you used! I have been trying to get itworking now for 2 days. It is very hard to get working!>If you want to give me a call in the office tomorrow, I can help you. We are Eastern time. My number is below. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389 f. 864-233-4344 "Ask me about Voice over IP." http://www.netxusa.com/
Thanks for the reply Sean and look forward for more. I do believe I have the 1204 configured ok and I am able to place outbound calls (from * to PSTN). I think my only hang up is some type of * extension config on incoming calls. * 101 type of stuff. I am still just learning. As a side note. I found (with help from the Mediatrix folks) that the getwalk feature was a great tool for configing the 1204. I just looked at the output for all the nat.0.x addresses to see where to plug in my nat.* address. That was my biggest hang up with the 1204. Now it is * config time. I really like the syslog feature on 1204. I have the logging cranked up to a level 5. Now I just have to figure out what all these messages mean. Sean P. Robertson wrote:>----- Original Message ----- >From: "Bob Knight" <bk@minusw.com> >To: <asterisk-users@lists.digium.com> >Sent: Wednesday, November 05, 2003 8:54 PM >Subject: Re: [Asterisk-Users] Mediatrix 1204 > > > > >>What a timely subject. I am setting here trying to bring up a 1204. >>I receive a sip invite from the 1204 but * is returning 404 extension >>not found. >> >>I am a newbie to * and am still fumbling around with config files. >>Could you please save a few of us a little time and share your * config >>files relating to the 1204. >> >>thanks in advance, bk..... >> >> >> >> > >Sure. I just saw another reply to this come in and he has a good start on >the Mediatix config steps. I will get together a list of some of the other >Mediatrix configuration parameters and the Asterisk relevant config files >that will work for you and email them to you tomorrow. > >Sean > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > > >
---------- Original Message ---------------------------------- From: "Sean P. Robertson" <spr@netxusa.com> Reply-To: asterisk-users@lists.digium.com Date: Wed, 5 Nov 2003 22:17:21 -0500>Sure. I just saw another reply to this come in and he has a good start on the Mediatix config steps. I will get together a list of some of the other Mediatrix configuration parameters and the Asterisk relevant config files that will work for you and email them to you tomorrow. > >SeanI just wanted to say thank you to Sean's He has been a great help over the phone to configure the Mediatrix 1204 setup for SIP. It works great and it does take a username and password! Now to the next question on the Mediatrix 1204. How can I save the setup as a script so I can run it when I get some more of them? Maybe I will just wait till they get there Web base application setup going! In the next 2 weeks or so! Again thank you very much! I will have to say Sean's the man!!!!!!---
Someone have the MIB for MEdiatrix 1204 version 2.4.10.68? thanks -- Almada Tres SA de CV Mitel Networks Eng. Gonzalo Gasca Meza Service Engineer 52+(55)53730570 Mexico City, Mexico
Actually im trying to set up a Mediatrix 1204 to place outgoing calls,i just cand do internal ones, i would like to know if someone could help me with this issue, i declared in sip.conf line1 to line4 for each 1204 port SIP.conf [100] ; My SIP agent type=friend ; This device takes and makes calls username=100 ; Username on device secret=100 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this voicemailbox has messages in it callerid="Gonzalo Gasca" <100> ; Caller ID [line1] type=friend ; This device takes and makes calls username=line1 ; Username on device host=110.10.200.10 ; This host is not on the same IP addr every time context=sip callerid="Line 1" <line1> ; Caller ID **************************************************************************************** extensions.conf **************************************************************************************** [sip] ignorepat => 9 exten => _9NNNNNNNN,1,Dial(SIP/line1) exten => :9NNNNNNNN,2,Congestion But it just put the box in busy and interchange rtp G711 packets with my client SJphone form sjlabs I would like a helping hand! -- _______________________________________________ Get your free email from http://www.hackermail.com Powered by Outblaze
The Mediatrix box will not registered with * as the user name and password
for sip are not yet implemented in their firmware.
All what you have to do is to protect the box from the internet (firewall)
and access is like:
exten => _1905XXXXXXX,1,Dial(SIP/${EXTEN}@IP_ADDR_OF_MEDIATRIX)
exten => _1905XXXXXXX,1,Congestion
This way you basically have a pool of 4 outgoing lines. You can however
route properly incoming calls.
I hope this will help you,
Regards,
Wojtek
----- Original Message -----
From: "Gonzalo Gasca" <ggasca@hackermail.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, June 07, 2004 9:45 PM
Subject: [Asterisk-Users] Mediatrix 1204
> Actually im trying to set up a Mediatrix 1204 to place outgoing calls,i
just cand do internal ones, i would like to know if someone could help me
with this issue, i declared in sip.conf line1 to line4 for each 1204
port>
> SIP.conf
>
> [100] ; My SIP agent
> type=friend ; This device takes and makes calls
> username=100 ; Username on device
> secret=100 ; Password for device
> host=dynamic ; This host is not on the same IP addr
every time> context=sip ; Inbound calls from this host go here
> mailbox=100 ; Activate the message waiting light if
this voicemailbox has messages in it> callerid="Gonzalo Gasca" <100> ; Caller ID
>
> [line1]
> type=friend ; This device takes and makes calls
> username=line1 ; Username on device
> host=110.10.200.10 ; This host is not on the same IP addr
every time> context=sip
> callerid="Line 1" <line1> ; Caller ID
>
>
****************************************************************************
************> extensions.conf
>
****************************************************************************
************>
> [sip]
> ignorepat => 9
> exten => _9NNNNNNNN,1,Dial(SIP/line1)
> exten => :9NNNNNNNN,2,Congestion
>
> But it just put the box in busy and interchange rtp G711 packets with my
client SJphone form sjlabs> I would like a helping hand!
> --
> _______________________________________________
> Get your free email from http://www.hackermail.com
>
> Powered by Outblaze
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Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204?
Bill Michaelson wrote:> Would anyone be kind enough to post a sip.conf fragment as a sample for > use with a Mediatrix 1204?Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian
I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp <florian@obsimref.com> wrote:> Bill Michaelson wrote: > > Would anyone be kind enough to post a sip.conf fragment as a sample for > > use with a Mediatrix 1204? > > Ours works with: > > [mtrix1] > type=peer > host=172.28.4.46 > mask=255.255.255.255 > context=in-mtrix1 > qualify=no > canreinvite=no > dtmfmode=inband > disallow=all > allow=ulaw > > > Best regards, > Florian > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >