[asterisk-users] Mediatrix 1204
Hi -
I've recently bought a mediatrix 1204 and have had a complete nightmare
getting it up and running with an [EMAIL PROTECTED] setup. I know this isn't a
mediatrix list but I'm at my wits end and the support with this product is
atrocious. (mine was even shipped with firmware that was incompatible with
the win32 software it came with so I wasted a day trying to work out why the
SNMP software wouldn't work )
I've finally managed to get incoming calls to work properly by getting it to
forward all calls to 4000 which is then passed on to the asterisk proxy and
treated as an inbound route that gets answered correctly.
The problem is then that when I place an outbound call through the gateway
it also forwards that back as well. It then uses each channel in order until
it fails as they're all busy.
The xml configuration file is at http://www.ascensus.co.uk/config.xml
The asterisk debug log is as below with my mobile replaced with mymobileno:
I've also attached sip.conf below. If anyone has any idea how to get this
thing to accept outgoing calls I would be very grateful of any input. All
the docs and howto's I've found state that it should 'just work' once the
inbound settings are working but I've not found that to be the case. The
settings are all defaults except the following:
Static IP address
Proxy server address
VAD on 711 disabled
Comfort noise disabled
AutomaticCallEnable yes
AutomaticCallTargetAddress 4000 (which is obviously the problem...)
Any help appreciated
Thanks
robbie
Sip.conf
[inbound]
type=friend
host=192.168.0.253
context=from-pstn
canreinvite=no
allow=ulaw
allow=alaw
asterisk1*CLI>
-- Executing Macro("SIP/4005-9d61", "dialout-trunk|7|mymobilenumber|")
in new stack
-- Executing GotoIf("SIP/4005-9d61", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/4005-9d61", "record-enable|4005|OUT") in new
stack
-- Executing GotoIf("SIP/4005-9d61", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/4005-9d61", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/4005-9d61", "RecEnable=RECORD-OUT/4005") in new
stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=4005
-- DBget: Value not found in database.
-- Executing SetVar("SIP/4005-9d61",
"CALLFILENAME=OUT4005-20070411-181258-1176311578.13302") in new stack
-- Executing Goto("SIP/4005-9d61", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("SIP/4005-9d61", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/4005-9d61", "NO RECORDING NEEDED") in new stack
-- Executing GotoIf("SIP/4005-9d61", "fooBgate:?7") in new stack
-- Executing SetCallerID("SIP/4005-9d61", "Bgate: Treatment (Large)
<4005>") in new stack
-- Executing Goto("SIP/4005-9d61", "9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("SIP/4005-9d61", "OUT_7") in new stack
-- Executing CheckGroup("SIP/4005-9d61", "") in new stack
-- Executing SetVar("SIP/4005-9d61", "DIAL_NUMBER=mymobilenumber") in
new stack
-- Executing SetVar("SIP/4005-9d61", "DIAL_TRUNK=7") in new stack
-- Executing AGI("SIP/4005-9d61", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/4005-9d61", "OUTNUM=mymobilenumber") in new
stack
-- Executing Cut("SIP/4005-9d61", "custom=OUT_7|:|1") in new stack
-- Executing GotoIf("SIP/4005-9d61", "0?19") in new stack
-- Executing Dial("SIP/4005-9d61", "SIP/inbound/mymobilenumber") in new
stack
We're at 192.168.0.254 port 12542
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
From: "Bgate: Treatment (Large)" ;tag=as5b17ec6a
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 11 Apr 2007 17:12:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1321 1321 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12542 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.253:5060
-- Called inbound/mymobilenumber
asterisk1*CLI>
Sip read:
SIP/2.0 100 Trying
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: "Bgate: Treatment (Large)" ;tag=as5b17ec6a
To: ;tag=2120bdca0a07567
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
Content-Length: 0
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
8 headers, 0 lines
asterisk1*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/
Re: [asterisk-users] Mediatrix 1204
I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp <[EMAIL PROTECTED]> 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 ___ --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
Re: [asterisk-users] 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 ___ --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
[asterisk-users] Mediatrix 1204
Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? ___ --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
[asterisk-users] Mediatrix 1204 and Asterisk 1.2.10
Hi Again, I thought I would repost in case some people missed it. I am currently set up with asterisk 1.2.10 on Mandriva 2006 using a Mediatrix 1204 as our media gateway. Everything works well during the day. However something occurs during the evening when the system is not in use that causes the communication between the Mediatrix and Asterisk server to break down. However, when I reboot the mediatrix everything is fine. I have recently upgraded the firmware on the mediatrix but that did not provide any solution. I apologize about the vague issue but the logs do not seem to be providing any clue as to what happens. I have a syslog server for the mediatrix and it is not showing anything out of the ordinary. Thanks Julian From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Thu, 3 Aug 2006 15:58:05 +Subject: [asterisk-users] Reboot Mediatrix Hi Everyone, Has anyone had an issue with the Mediatrix and Asterisk where you need to reboot the mediatrix every morning? When I try to place calls or send calls in I am receiving a busy signal. I look at the sip show channels and all the channels seem to be used by the mediatrix. When I reboot everything is fine. Thanks Julian ___ --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
[asterisk-users] Mediatrix 1204 and Asterisk 1.2.9 stops working intermittently
I have a mediatrix 1204 which is connecting with asterisk 1.2.9. I have created a howto as well, but I am now encountering one interesting problem that did not occur with asterisk 1.0.8. Every so often the mediatrix will not handoff a call to the asterisk box, until I change the login name on both the mediatrix box and asterisk for sip authentication. After that everything runs fine..for a while. I have scanned the logs but the only error I see every once in a while is "chan_sip.c:10988 in handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 1877XXX in context default" 1877XXX is our 800 number Has anyone had a similar issue? If I were to add a hint for the mediatrix how would I do this? Thanks Julian > Date: Thu, 13 Jul 2006 12:38:31 -0400> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?> > Ditto here.. we were running 1.2.6.. decided to upgrade to 1.2.9.1...> crash crash crash crash... so we downgraded back to 1.2.6 and have> been up for weeks at a time now without issues.> > > > On 7/12/06, j <[EMAIL PROTECTED]> wrote:> > I personally have had some issues with 1.2.9.1 in production and had to> > revert to an older version.> > We are using 1.2.6 which has proven to be pretty stable.> >> > Others might have different experiences.> >> > j> >> > On Wed, 2006-07-12 at 15:31 +0200, Andrea Spadaccini wrote:> > > Hello,> > > I need to install Asterisk on a test machine that will soon become a> > > production environment.> > >> > > Do you think that 1.2.9.1 is reliable? I read some posts that say it> > > isn't as good as the previous versions. Should I install 1.2.8 or> > > 1.2.7.1?> > >> > > Please give me an advice!> > > Thanks in advance,> >> >> > ___> > --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> >> ___> --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 ___ --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
[Asterisk-Users] Mediatrix 1204 and Asterisk
Hi Everyone, I am new to Asterisk but I have found that quite a few people have implemented it with the Mediatrix 1204. Does anyone know of a wiki or place where there is good documentation regarding this configuration? Thanks Julian___ --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
Re: [Asterisk-Users] Mediatrix 1204 help please.
On 12/16/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > OK we need some help in setting up a good wiki-info page for setting up the > > Mediatrix > 1204 to work with asterisk. If anyone has > > set these unit's up and have them working please post your settings here so > > we can > create a page on the wiki. These unit's are > > being sold to be used via sip format with asterisk and there is no real > > information > on getting them working. At present there one > > of the worst I have run into to get correctly working. These are very > > expensive and > some of us can't afford to send them back for > > a restocking fee. > > > > If someone working with Mediatrix has a white paper on getting these unit's > > working > please let us know the link for it. It would > > be very helpful for many asterisk users. > > > > If you search the -users archives, you'll see where a couple of people > have made them work. I believe there was at least one posting reflecting > a working config. > > I did an eval on the 1204 in early 2004, but did not care for the way > it interfaced with asterisk. The 1204 was really intended to interoperate > with the 1104 as a toll bypass box. > > I was able to make it work and the audio was excellent with no echo > whatsoever. Key items (in early 2004) included: > - the 1204 does not have any sip register functions. One must configure it > (and asterisk) to work with static IP addresses (instead of relying on > the registration process). > - calls from asterisk "to" (or through) the 1204 are treated as a group > and the 1204 chooses the first available pstn port for all calls. If you > want to direct a call to a specific port, one has to jump through hoops > to force a CallerID (from asterisk) and then program 1204 to look for the > callerid (which is then used to match a port number). Not cool. > - programming the 1204 could only be done via snmp, and the snmp facility > provided only ran on Windows. Each firmware upgrade to the 1204 required > a new snmp implementation as the mib variables constantly changed. The > snmp community string (eg, password) could not be changed from public, > therefore exposing the 1204 to the internet would be a major security > risk. (If you know snmp extemely well, you can use the mib definitions > within a linux system to program the box, but you better be very good > at snmp to do that.) They now support a limited set of things that could be programmed via http. > - Support for the box is only offered through resellers, and their typical > resellers are those firms reselling traditional pbx's. A fair number of > those don't have a clue what voip is about and even fewer can spell > asterisk. I thought they teach you how to spell asterisk in 4th grade math? :) > - All firmware upgrades are chargable regardless of what problem might be > found. The upgrade charge was very high (something like $500 in 2004). > > Given the above (in 2004), the risk associated with using the 1204 was > far to great and I returned the unit for full credit. (The eval was arranged > through Mediaxtix sales rep even though the unit came from a reseller.) > > I've not touched or seen the 1204 since early 2004, so can't help any more > then what is stated above. The product may have improved since then, but > I don't have clue what might have changed. > > I have set it up many times now, but each time I have to do that, it takes around an hour, which is a very long time for me compared to an Adit or Sipura, but because of the quality of the box (sound is very good) I still use it. Each time I do it I just go thru the SNMP settings one by one, and set/change what I think/know is needed, then test it to make sure it works as I want. It would take me an extra hour to write it down and post on the wiki, but I'll try to do it next time anyhow, that way it might save me some time in the future as well :) ___ --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
Re: [Asterisk-Users] Mediatrix 1204 help please.
> OK we need some help in setting up a good wiki-info page for setting up the > Mediatrix 1204 to work with asterisk. If anyone has > set these unit's up and have them working please post your settings here so > we can create a page on the wiki. These unit's are > being sold to be used via sip format with asterisk and there is no real > information on getting them working. At present there one > of the worst I have run into to get correctly working. These are very > expensive and some of us can't afford to send them back for > a restocking fee. > > If someone working with Mediatrix has a white paper on getting these unit's > working please let us know the link for it. It would > be very helpful for many asterisk users. > If you search the -users archives, you'll see where a couple of people have made them work. I believe there was at least one posting reflecting a working config. I did an eval on the 1204 in early 2004, but did not care for the way it interfaced with asterisk. The 1204 was really intended to interoperate with the 1104 as a toll bypass box. I was able to make it work and the audio was excellent with no echo whatsoever. Key items (in early 2004) included: - the 1204 does not have any sip register functions. One must configure it (and asterisk) to work with static IP addresses (instead of relying on the registration process). - calls from asterisk "to" (or through) the 1204 are treated as a group and the 1204 chooses the first available pstn port for all calls. If you want to direct a call to a specific port, one has to jump through hoops to force a CallerID (from asterisk) and then program 1204 to look for the callerid (which is then used to match a port number). Not cool. - programming the 1204 could only be done via snmp, and the snmp facility provided only ran on Windows. Each firmware upgrade to the 1204 required a new snmp implementation as the mib variables constantly changed. The snmp community string (eg, password) could not be changed from public, therefore exposing the 1204 to the internet would be a major security risk. (If you know snmp extemely well, you can use the mib definitions within a linux system to program the box, but you better be very good at snmp to do that.) - Support for the box is only offered through resellers, and their typical resellers are those firms reselling traditional pbx's. A fair number of those don't have a clue what voip is about and even fewer can spell asterisk. - All firmware upgrades are chargable regardless of what problem might be found. The upgrade charge was very high (something like $500 in 2004). Given the above (in 2004), the risk associated with using the 1204 was far to great and I returned the unit for full credit. (The eval was arranged through Mediaxtix sales rep even though the unit came from a reseller.) I've not touched or seen the 1204 since early 2004, so can't help any more then what is stated above. The product may have improved since then, but I don't have clue what might have changed. ___ --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
RE: [Asterisk-Users] Mediatrix 1204 help please.
Ariel, There are some notes in the list archives about getting them going. -Nate -Original Message- From: Ariel Batista [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 9:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mediatrix 1204 help please. OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many asterisk users. ___ --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
[Asterisk-Users] Mediatrix 1204 help please.
OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many asterisk users. ___ --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
Re: [Asterisk-Users] Mediatrix 1204 and Asterisk
Let's say the incoming call you wanted to route came into the [incoming] context: [context] exten => s,1,Answer exten => s,2,Playback(file) exten => s,3,Hangup exten => s/8005551212,1,Answer exten => s/8005551212,2,Playback(specialfile) exten => s/8005551212,3,Hangup If the callerid of the caller connecting is exactly 8005551212, the second block is matched. otherwise, the first block, without the slash and the callerid string, would be the default match-all extension. I believe (check the wiki) there is a syntax that combines the callerid matching with the wildcard matching: exten => s/_800.,1,Answer This would match any callerid string beginning with 800. Please look over the wiki to make sure the syntax for this example is correct before you try it. Moj Shad Mortazavi wrote: Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo <[EMAIL PROTECTED]> Office Manger, Horan & Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 and Asterisk
Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 setup
Hi Sean: My name is Maribel Ronquillo, right now I’m working with VoIP products like snom IP Phones, sixpert, sipura and mediatrix. I have had some doubts about the setup of the last product, is a Mediatrix 1204 gateway, and I don’t know how to setup de SPEED DIAL of it, I have the administrator Manual but I doesn’t help much. Do you have any other information about it that could help me? Thanks Maribel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 caller ID
Has anyone been able to get one of these boxes to pass on the caller ID to Asterisk? When an incoming call comes through, softphone and hardphone only display "pstnline1" or whatever you name the ports in the device. I'm looking at having about 8 incoming lines and I was thinking 2 of these Mediatrix boxes would do the trick. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mediatrix 1204 Help
How do metratrix board wirk with asterisk? Any scenarios? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Time Bandit |Sent: Martes, 03 de Mayo de 2005 09:11 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Mediatrix 1204 Help | |> I just got Mediatrix 1204 from ebay, but it is missing CD that |> conmtain the software and drivers, I am wondering if anybody knows |> where I could downloaded from. |Have you tried http://www.mediatrix.com/ ? |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
> >> > >> I just got Mediatrix 1204 from ebay, but it is missing CD that > conmtain the software and > > drivers, I am wondering if > > >> anybody knows where I could downloaded from. > >> > > > >The firmware is not openly available. Mediatrix approach is to "charge" > >customers for every release they generate, and they only do that > >through approved resellers. If you know a company that resells their > >products, you might be able to twist their arm, but I'd guess they > >aren't going to give it away. (That's probably why it was being sold > >on eBay in the first place.) > > >You will need the firmware that runs on the box (be sure to get the sip > >version), and you'll need the Windows-only snmp management software > >to configure the thing. Each firmware version has a specific snmp > >management package intended to be used with the firmware. You'll need > >both (matching) to accomplish anything as there is no telnet or web > >interface. > > No no no. Screw windows. All you need is the mib files and mbrowse. > SNMP makes remote admin of these boxes a piece of cake. Much faster > then a web browser. Once you figure out what you are doing, then you > can just config and admin it with simple shell scrips, or if your a > hack like me, c code. You can even use SNMP to monitor the PSTN > line status. Way cool stuff and these boxes just run forever. For those of us that are somewhat heavy into snmp, I'd agree. But a large percentage of asterisk users don't ever deal with it or even know what it is. I'd agree on the stability of the box. Very nice, good echo cancellation, etc. Less then satisfactory in how they deal with sip (eg, registration), security, etc. For internal use, no problem; for external, I'd never expose it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
Message: 16 Date: Tue, 3 May 2005 09:12:13 -0600 From: Rich Adamson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Mediatrix 1204 Help To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 >> >> I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if >> anybody knows where I could downloaded from. >> >The firmware is not openly available. Mediatrix approach is to "charge" >customers for every release they generate, and they only do that >through approved resellers. If you know a company that resells their >products, you might be able to twist their arm, but I'd guess they >aren't going to give it away. (That's probably why it was being sold >on eBay in the first place.) >You will need the firmware that runs on the box (be sure to get the sip >version), and you'll need the Windows-only snmp management software >to configure the thing. Each firmware version has a specific snmp >management package intended to be used with the firmware. You'll need >both (matching) to accomplish anything as there is no telnet or web >interface. No no no. Screw windows. All you need is the mib files and mbrowse. SNMP makes remote admin of these boxes a piece of cake. Much faster then a web browser. Once you figure out what you are doing, then you can just config and admin it with simple shell scrips, or if your a hack like me, c code. You can even use SNMP to monitor the PSTN line status. Way cool stuff and these boxes just run forever. If you are ready to give up on the boxes and want to dump them at a good price, just let me know. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
Hi Time, yes, I did but nothing on their website Cheers, Ellafi Time Bandit <[EMAIL PROTECTED]> wrote: > I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the> software and drivers, I am wondering if anybody knows where I could> downloaded from. Have you tried http://www.mediatrix.com/ ?___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersKeep smiling life is too shortGo for the best and expect the worst.Do not be afraid to take a chances in life, as life it self is a chance.Do not fear anybody but God and you will be protectedfrom anything that does not fear him.Eat good and dress good and believe in God you will live healthy.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
> > I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain > the software and drivers, I am wondering if > anybody knows where I could downloaded from. > The firmware is not openly available. Mediatrix approach is to "charge" customers for every release they generate, and they only do that through approved resellers. If you know a company that resells their products, you might be able to twist their arm, but I'd guess they aren't going to give it away. (That's probably why it was being sold on eBay in the first place.) You will need the firmware that runs on the box (be sure to get the sip version), and you'll need the Windows-only snmp management software to configure the thing. Each firmware version has a specific snmp management package intended to be used with the firmware. You'll need both (matching) to accomplish anything as there is no telnet or web interface. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
> I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the > software and drivers, I am wondering if anybody knows where I could > downloaded from. Have you tried http://www.mediatrix.com/ ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 Help
Hi, I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. Please help, Thank you. Do you Yahoo!? Make Yahoo! your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 Help
Hi, I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. Please help, Thank you. Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 Help
Hi, I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. Please help, Thank you. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 DialPlan and Delay
Hi everybody, I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing. The problem here is the delay. When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call. For OUTGOING My Dialplan for the Mediatrix box is the following, here at Mexico we use 8 digits for local calls. ([1-9]xxx|01xx||060|0xx) I have verified that inmediatly after I dial from my IP phone, the in-use light turns on in Mediatrix but the call is not pass until the 4 seconds timer expires. I have tried disabling the Dial plan but it didnt help Form Mediatrix documentation The Timer is set to 4 seconds. It can be used to indicate that if users have not dialed a digit for 4 seconds, it is likely that they have finished dialing and the gateway can make the call. A Dial Map for this could be: [2-9]xxT FOR INCOMING The same 4 seconds delay after the call is sent to Asterisk. The problem here, is that despite we answer or not the call, once the call is sent to Mediatrix, the calling party hear 2 ring-back tones generated by Mediatrix, then the ringback for Asterisk Once the call is passed to Asterisk and starts ringing, if we call from a cell phone,home or office the call is marked as answered and the call timer starts no matter if is answered or not. Any ideas? I have tried sending the # at the end with no success. Thanks! Do you Yahoo!? Send a seasonal email greeting and help others. Do good.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204, where I can get the last firmware and mib file ?
You have to get it from your reseller. Mediatrix does not provide free upgrades, therefore your reseller will charge a fee for it. > Where I can get the last firmware and mib file for the Mediatrix 1204 ? > > Kind regards, > > Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 - Error: Operation not permitted
[EMAIL PROTECTED] wrote: When I try to make a call to PSTN via Mediatrix 1204 I received the error below: Aug 7 21:01:48 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x482082ec (len 430) to 192.168.199.5 returned -1: Operation not permitted There are here anyone that knows what I can do to correct it ? Crank up the syslog debug level to 5 on the 1204. Even if you do not have a syslogd running (but you should) you can still read all the ascii messages with ethereal. This will provide pretty good debug messages. When you are done debugging, I would suggest dropping the level back down to 4. It gets a little verbose. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 - Error: Operation not permitted
When I try to make a call to PSTN via Mediatrix 1204 I received the error below: Aug 7 21:01:48 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x482082ec (len 430) to 192.168.199.5 returned -1: Operation not permitted There are here anyone that knows what I can do to correct it ? Kind regards, Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204, where I can get the last firmware and mib file ?
Where I can get the last firmware and mib file for the Mediatrix 1204 ? Kind regards, Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mediatrix 1204 hysteria
Jair Martinez wrote: I read that some of you installed mediatrix devices with a SIP server and it worked OK. Could you please tell me which SIP server you used, and how did you configure it on the 1204? The SIP server is called something like asterisk. The only problem I had with 1204's was having to use a damn windows box for config. But now that I have it working with mbrowse on linux, the universe is in balance again. My office is back to a totally microsoft free environment. I did have to use the windows box to grab the mib files off the 1024's cd. For some reason I can not read that cd on a linux box. Anyone know why? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 has not given me any feedback for my problem in more than a month. I am feeling despertae. I am installing a Mediatrix 1204 with a sip server and a PBX that manage all my PST lines. The main problem that I have is with the outgoing calls from the 1204. I can only make one call and then after the outgoing call ends, I have to reset the 1204 in order to call again. Most of the calls (85%) we make from external lines (PST) work. I am using a mediatrix 1204 and ondo sip server in one of our offices and a mediatrix 2102 in the other office. Have you ever experienced this problem before? I read that some of you installed mediatrix devices with a SIP server and it worked OK. Could you please tell me which SIP server you used, and how did you configure it on the 1204? I would really appreciate if you could help me with this. Jair
[Asterisk-Users] Mediatrix 1204 Incoming calls
I would like to know if someone could help me when i recieved an incoming call on a Meditarix 1204 how to redirect the call? And the configuration i need? _Thanks -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 Mibs
I´d like to know if someone could help me with this issue: Anybody there have the Mibs for .68 ver in 1204 ? or meavy the .cfg file. I found mediatrix box so hard to configure. -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 configuration
I really can't help you with this. The last time I touched the 1204 was about three months ago, and it used v2.4.9.57 at that time. I did use the unit manager at the time. When I did have the 1204, I had to upgrade the code as our reseller could not resolve documented problems. The mib files "have to match" the version of software you're trying to use. This single upgrade included a completely different set of mibs from what had been in use, and even the reseller specifically stated that I had to keep the mib files and firmware in sync. I did not keep any notes, etc, after returning the unit to the reseller. Rich > Thank u very much Rich! > > I did what u suggested me, but im still having problems with the Mediatrix, actually > i dont have the MIbs for version 2.4.10.68, i tested 1204 with a different SIP server called 3050 from Mitel www.mkcnetworks.com and it worked ok. Could help me with the mediatrix configuration? Have u used Unit Manager for Mediatrix to configure it? > Could you help me with .cfg file? > > Thanks a lot! > - Original Message - > > Gonzalo, > > > > > i would like if some could help me with a * and Mediatrix configuration... > > > i have this in my extensions.conf file > > > > > > [outbound] > > > ignorepat => 9 > > > exten => _901,1,Dial(SIP/[EMAIL PROTECTED]) > > > exten => _901,2,Congestion > > > exten => _9020,1,Dial(SIP/[EMAIL PROTECTED]) > > > exten => _9020,2,Congestion > > > > > > and this in my sip file > > > > > > [Mediatrix] > > > type=peer > > > host=110.10.200.10 > > > mask=255.255.255.255 > > > qualify=yes > > > canreinvite=no > > > disallow=g729 > > > > You might want to use google to find several postings regarding the 1204 > > over the last six months or so. Search for "mediatrix sip dial" and/or > > "mediatrix 1204" for several examples. > > > > Here's a couple: > > http://lists.digium.com/pipermail/asterisk-users/2004-February/036015.html > > http://lists.digium.com/pipermail/asterisk-users/2004-February/036833.html > > > > It will probably be much easier for you (to start with) to forget about > > the mediatrix context (above) for outbound calls, and just do something > > like: > > exten => _9.,1,Dial,SIP/[EMAIL PROTECTED] > > which worked in the past. Then mess with the mediatrix configuration until > > the above dials out correctly with only a single pstn line connected. > > > > The mediatrix will strip the first digit dialed unless you change the > > country code which is undocumented by mediatrix (which can also be found > > via google). > > > > Finding the appropriate mediatrix snmp mib variables to support your > > environment can be somewhat difficult since the terminology used by > > Mediatrix is somewhat different from asterisk. > > > > Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 configuration
Thank u very much Rich! I did what u suggested me, but im still having problems with the Mediatrix, actually i dont have the MIbs for version 2.4.10.68, i tested 1204 with a different SIP server called 3050 from Mitel www.mkcnetworks.com and it worked ok. Could help me with the mediatrix configuration? Have u used Unit Manager for Mediatrix to configure it? Could you help me with .cfg file? Thanks a lot! - Original Message - From: Rich Adamson <[EMAIL PROTECTED]> Date: Tue, 15 Jun 2004 06:23:04 -0600 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Mediatrix 1204 configuration > Gonzalo, > > > i would like if some could help me with a * and Mediatrix configuration... > > i have this in my extensions.conf file > > > > [outbound] > > ignorepat => 9 > > exten => _901,1,Dial(SIP/[EMAIL PROTECTED]) > > exten => _901,2,Congestion > > exten => _9020,1,Dial(SIP/[EMAIL PROTECTED]) > > exten => _9020,2,Congestion > > > > and this in my sip file > > > > [Mediatrix] > > type=peer > > host=110.10.200.10 > > mask=255.255.255.255 > > qualify=yes > > canreinvite=no > > disallow=g729 > > You might want to use google to find several postings regarding the 1204 > over the last six months or so. Search for "mediatrix sip dial" and/or > "mediatrix 1204" for several examples. > > Here's a couple: > http://lists.digium.com/pipermail/asterisk-users/2004-February/036015.html > http://lists.digium.com/pipermail/asterisk-users/2004-February/036833.html > > It will probably be much easier for you (to start with) to forget about > the mediatrix context (above) for outbound calls, and just do something > like: > exten => _9.,1,Dial,SIP/[EMAIL PROTECTED] > which worked in the past. Then mess with the mediatrix configuration until > the above dials out correctly with only a single pstn line connected. > > The mediatrix will strip the first digit dialed unless you change the > country code which is undocumented by mediatrix (which can also be found > via google). > > Finding the appropriate mediatrix snmp mib variables to support your > environment can be somewhat difficult since the terminology used by > Mediatrix is somewhat different from asterisk. > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 configuration
Gonzalo, > i would like if some could help me with a * and Mediatrix configuration... > i have this in my extensions.conf file > > [outbound] > ignorepat => 9 > exten => _901,1,Dial(SIP/[EMAIL PROTECTED]) > exten => _901,2,Congestion > exten => _9020,1,Dial(SIP/[EMAIL PROTECTED]) > exten => _9020,2,Congestion > > and this in my sip file > > [Mediatrix] > type=peer > host=110.10.200.10 > mask=255.255.255.255 > qualify=yes > canreinvite=no > disallow=g729 You might want to use google to find several postings regarding the 1204 over the last six months or so. Search for "mediatrix sip dial" and/or "mediatrix 1204" for several examples. Here's a couple: http://lists.digium.com/pipermail/asterisk-users/2004-February/036015.html http://lists.digium.com/pipermail/asterisk-users/2004-February/036833.html It will probably be much easier for you (to start with) to forget about the mediatrix context (above) for outbound calls, and just do something like: exten => _9.,1,Dial,SIP/[EMAIL PROTECTED] which worked in the past. Then mess with the mediatrix configuration until the above dials out correctly with only a single pstn line connected. The mediatrix will strip the first digit dialed unless you change the country code which is undocumented by mediatrix (which can also be found via google). Finding the appropriate mediatrix snmp mib variables to support your environment can be somewhat difficult since the terminology used by Mediatrix is somewhat different from asterisk. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 configuration
i would like if some could help me with a * and Mediatrix configuration... i have this in my extensions.conf file [outbound] ignorepat => 9 exten => _901,1,Dial(SIP/[EMAIL PROTECTED]) exten => _901,2,Congestion exten => _9020,1,Dial(SIP/[EMAIL PROTECTED]) exten => _9020,2,Congestion and this in my sip file [Mediatrix] type=peer host=110.10.200.10 mask=255.255.255.255 qualify=yes canreinvite=no disallow=g729 when i dial via SJlabs phone i see the online led turn on and the 1204 via asterisk start to interchange rtp packets but it never send the digits to the pstn. could someone help me with the mediatrix 1204 configuration? via their .cfg file? uisng UMN Thanks -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 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 Tryc" <[EMAIL PROTECTED]> Date: Mon, 7 Jun 2004 21:59:43 -0400 To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Mediatrix 1204 > 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 => _1905XXX,1,Dial(SIP/[EMAIL PROTECTED]) > exten => _1905XXX,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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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" ; Caller ID > > > > > > > > extensions.conf > > > > > > > > [sip] > > ignorepat => 9 > > exten => _9,1,Dial(SIP/line1) > > exten => :9,2,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 > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
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 => _1905XXX,1,Dial(SIP/[EMAIL PROTECTED]) exten => _1905XXX,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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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" ; Caller ID > > > extensions.conf > > > [sip] > ignorepat => 9 > exten => _9,1,Dial(SIP/line1) > exten => :9,2,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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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" ; Caller ID extensions.conf [sip] ignorepat => 9 exten => _9,1,Dial(SIP/line1) exten => :9,2,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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
Their current firmware doesn't allow to write to the section for SIP registration. I am able to communicate with it by dialing [EMAIL PROTECTED]. Also, you have to protect this box with Firewall otherwise the whole world will be able to call through it. Regards, Wojtek - Original Message - From: Dawid Mielnik To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 9:40 AM Subject: RE: [Asterisk-Users] Mediatrix 1204 (4x FXO) And what problem do you have with registering ? Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might reference that, configuring 1204 should be very similar to that of 1104. Regards, Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech TrycSent: Thursday, May 06, 2004 5:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 1204 (4x FXO) I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
> > Don't know how far you've tried to take the 1204 in terms of functions, > but we did the same thing over a two month period and found: > > 1. handling outbound calls on a "per pstn line" basis (eg, directing > certain calls to certain pstn lines) is very non-standard and subject > to future failures as code changes happen in * and the 1204. Correct, but I don't need to have access on per channel basis. > 2. ring-cadence detect is done on the first ring after the 1204 reboot > and applied to all four ports. If the pstn lines happen to come from > different Central Offices (with slightly different cadences), callerid > and other such timing sensitive functions will fail. I believe that you can actually change that, you have to specify time in ms not a number of rings. > 3. security is less then acceptable. If the 1204 is exposed to the > Internet, anyone can make calls, change settings, etc. Correct, but in real production wouldn't you keep it behind the Firewall? > 4. the per-port cost is substantially higher then many other products > "if" you consider the cost of keeping the firmware reasonably current > as standards evolve. Yes, but SIP connectivity (instead of PCI) adds lots of flexibility > 5. the box does not follow published sip standards; only selected pieces. I am sure that they will release new firmware with better support for SIP > 6. diagnosing problems and monitoring operational functions in a real-world > production environment is less then acceptable. Agreed > 7. support is limited to whatever your reseller provides, which is less > then acceptable if your reseller is not familiar with *. This is reality of the 21st century :) > > We also found the voice quality to be very good, echo cancellation was > good, etc. With relatively easy firmware tweeks to interoperate with * > and standards better, it would be a nice pstn interface; however, they > seem to not have any interest in going there. :) Regards, Wojtek > > Rich > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mediatrix 1204 (4x FXO)
The 1204 does not have the software routines implemented for "register". Their approach is the 1104 "registers" with the 1204. > And what problem do you have with registering ? > Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you > might reference that, configuring 1204 should be very > similar to that of 1104. > > Regards, > Dave > > -Original Message- > I have successfully implemented 1204 in semi production environment. Just want > to share that it works very well, through the firewall > (NATed). > Unfortunately, it can not register with the server (and authenticate) but > otherwise everything is fine. The audio quality is very good. > Regards, > Wojtek ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mediatrix 1204 (4x FXO)
And what problem do you have with registering ? Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might reference that, configuring 1204 should be very similar to that of 1104. Regards, Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech TrycSent: Thursday, May 06, 2004 5:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 1204 (4x FXO) I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
> Don't know how far you've tried to take the 1204 in terms of functions, > but we did the same thing over a two month period and found: > There is also an acknowledged bug that is a showstopper for us: configuration over DHCP fails, because the vendor code for outbound proxy is not recognised by the device. So each device must be manually reconfigured if, for example, the subnet address of the Asterisk box changes - it can't just be rebooted, fetch its DHCP payload, and reconfigure itself. All of the other vendor codes we've tried work. It seems like such a small bug, but none of the Mediatrix firmware updates cover it. Also, we couldn't get ringback to work right, despite a correct indications.conf file - if someone calls in and dialed an extension, then the ringing they hear is very choppy and messed up. I don't suppose anyone on the list has tested any other FXO gateways that are comparable to the 1204? Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 FXO GW ring cadence question
I'm still in a test mode with a new Mediatrix 1204 fxo gateway, and been having an issue with the 1204 properly detecting callerid. Two pstn lines installed, both with callerid. One pstn line rings with a standard US ring (long ring) Second pstn line is a CO Centrex and rings with a long+short ring It appears the 1204 senses and "sets" the ring cadence used for the CO centrex line (long + short ring), and looks for the callerid after that second ring. It then apparently uses that setting for all four lines, as the first pstn line (long ring) never accepts the callerid once the CO centrex line has rung. (After a reboot, the 1204 properly detects the callerid. But, after the CO centrex rings, it never detects callerid on the normal pstn line again.) Have any of you 1204 users bumped into that before? Any work arounds? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip g/w now working
I've had one of these things working for ages, although I never set it up to
select which port to use on outgoing lines. I overcame the first-digit
stripping by telling the 1204 to prefix outgoing calls not in my area code. I
seem to remember it stripping leading zeroes (as in 011 for international
calls). I should try your undocumented feature.
The other thing I've had troubles with is provisioning it via DHCP. All of the
DHCP key-value pairs are recognised except the one for outgoing proxy. It's
very annoying, and seems to be a firmware bug. So I've configured the gateway
to use a static IP.
Anyway, once set up, it seems to work okay, though it of course suffers from
the same hangup detection problems that afflict all users of loop start.
Thanks for the config tip to manually select outgoing ports; that could be
handy.
Christian
On Wednesday 11 February 2004 14:51, Rich Adamson wrote:
> For those that might have the Mediatrix 1204 4-port FXO sip gateway or
> for those that might have an interest, finally got it to work the way
> one would expect when interconnecting to analog pstn lines.
>
> Configuring the box for incoming calls was rather easy and worked
> shortly after installing the box.
>
> Configuring it for outgoing pstn calls has been at least a two week effort
> interacting with the reseller multiple times. The issues:
>
> Port Selection:
> ---
> The 1204 does not provide any documented method to "select" which of the
> four ports will be used for outgoing calls. The manufacturer assumes all
> four ports are the equivalent of a trunk group.
> Fix:
> In extensions.conf, add something like:
> exten => _6X.,1,SETCIDNUM()
> exten => _6X.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _6X.,3,Congestion
> and in the 1204:
> set gatewayPort1NetToPstnSourceFilter =
> Since the callerid that is set in asterisk never gets forward out the pstn
> line, the above mechanism works fine for selecting port 1. (Use , ,
> for the remaining ports.)
>
> Outbound calls dropping first digit:
>
> The 1204 automatically drops the "1" when calling any long distance call
> such as 1-800-555-1212.
> Fix:
> on the 1204, set countryCountryCode = 2
> This is an undocumented item, but essentially stops the 1204 from stripping
> leading digits.
>
> Summary:
>
> The limited testing conducted thus far indicates the 1204 is working very
> well. There is no noticeable echo at any time. Seems to work very well with
> canreinvite=yes although I've not tried it with a remote nat phone.
>
> One of the nice things about the box is you can locate it at your demarc
> and not have to provide 2-wire pstn connections to the asterisk system.
>
> Rich
>
>
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[Asterisk-Users] Mediatrix 1204 sip g/w now working
For those that might have the Mediatrix 1204 4-port FXO sip gateway or
for those that might have an interest, finally got it to work the way
one would expect when interconnecting to analog pstn lines.
Configuring the box for incoming calls was rather easy and worked
shortly after installing the box.
Configuring it for outgoing pstn calls has been at least a two week effort
interacting with the reseller multiple times. The issues:
Port Selection:
---
The 1204 does not provide any documented method to "select" which of the
four ports will be used for outgoing calls. The manufacturer assumes all
four ports are the equivalent of a trunk group.
Fix:
In extensions.conf, add something like:
exten => _6X.,1,SETCIDNUM()
exten => _6X.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _6X.,3,Congestion
and in the 1204:
set gatewayPort1NetToPstnSourceFilter =
Since the callerid that is set in asterisk never gets forward out the pstn
line, the above mechanism works fine for selecting port 1. (Use , ,
for the remaining ports.)
Outbound calls dropping first digit:
The 1204 automatically drops the "1" when calling any long distance call
such as 1-800-555-1212.
Fix:
on the 1204, set countryCountryCode = 2
This is an undocumented item, but essentially stops the 1204 from stripping
leading digits.
Summary:
The limited testing conducted thus far indicates the 1204 is working very
well. There is no noticeable echo at any time. Seems to work very well with
canreinvite=yes although I've not tried it with a remote nat phone.
One of the nice things about the box is you can locate it at your demarc
and not have to provide 2-wire pstn connections to the asterisk system.
Rich
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Re: [Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review
Rich Adamson wrote: Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. Trouble shooting is limited to the SNMP manager only. The manager can be used to view configuration data, however needed dynamic operational statistics are limited to mib2 definitions only. For example, when trying to determine the souce of choppy MOH sound, I wanted to check the Ethernet port speed. There was no mib variable defined for this purpose. I found the syslog feature pretty niffty. You crank the syslog up to level 5 and get a lot of info. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review
Product Review
Mediatrix 1204 4-Port SIP FXO Gateway
Firmware: v2.4.10.69 - US Version
US Retail: ~$750, Street Price: ~$450.
The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks
and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn
lines in either Loop Start or Ground Start mode, handles incoming CallerID,
and generates either Dial Tone (back towards the incoming pstn caller) or
redirects the call to a specific pre-programmed sip proxy extension. The
1204 can only be programmed through an SNMP (Simple Network Management
Protocol) manager. A Windows-based SNMP manager is supplied with the unit;
but no Unix-based manager. (And, no telnet, no web.)
To use the 1204 with Asterisk, each of the four pstn lines "must" be redirected
to an Asterisk extension. In this eval case, port 1 was redirected to x3091,
port 2 to x3092, etc. The 1204 detects the incoming call, and about midway
through the second ring, sends a sip Invite with the CallerID (if available)
to the defined sip proxy server (Asterisk). (After Asterisk completes the call
to another sip phone and the pstn caller hangs up, the Asterisk sip phone
will continue to ring for at least two-to-four additional ringing cycles.)
The firmware version tested did not support the sip "register" function even
though parameters were provided to enter the IP address of a registrar. As
a result, no userid/passwords or other security features are available.
Sip.conf security entries are limited to "host=" and context=
". All incoming port 1 calls are directed to an extension.conf
construct similar to exten => 3091,1,Goto(my-ivr) contained within the
section.
Again since the 1204 does not support the sip "register" function, outgoing
pstn calls from Asterisk can only be sent to the 1204 with commands similar to:
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
where the 1204 selects one of the non-busy four pstn ports at random to initiate
the pstn call. The reviewer could not find a way to direct specific Asterisk
calls to specific 1204 ports, and believes Mediatrix needs to fully implement
the "register" command on a per-port basis to aid in this requirement.
Echo cancellation and transmission levels were excellent on all inbound and
outbound calls. Ringback tone and Music on Hold (MOH) were extremely choppy
until the Port1DspVoiceActivityDetection = 0 parameter disabled this function.
NAT is supported according to the documentation, however I did not test this
to see if it actually worked. The standard rtp redirection (canreinvite=yes)
appeared to function properly.
As mentioned, the only way to configure the 1204 is via an SNMP Manager. There
is no way to change/secure the SNMP-v1 community string, therefore this box
should never be exposed to the Internet. The *.pdf documentation files are
very verbose and good (Admin = 196 pages); however there are no references to
Asterisk, leaving the reader to guess at how some functions actually
inter-operate, etc.
Opinion:
It would appear the 1204 is oriented to inter-operate with another 1204 across
the Internet, creating essentially a virtual pstn line extension to some
distant point. The box is available with either H-323 or SIP images, but not
both. One can only assume the incomplete SIP implementation is the result of
retrofitting the 323-based box into the SIP world. Since much of the *.pdf
documentation and files were dated March/April 2003, it does not appear
that SIP advancement is high on Mediatrix's list of priorities. Support for
the unit is limited by Mediatrix to "resellers only", therefore obtaining
any relevant support data in a timely manner is 100% dependent on how well
your reseller will support you.
Trouble shooting is limited to the SNMP manager only. The manager can be used
to view configuration data, however needed dynamic operational statistics are
limited to mib2 definitions only. For example, when trying to determine the
souce of choppy MOH sound, I wanted to check the Ethernet port speed. There
was no mib variable defined for this purpose.
Overall, the 1204 functioned very well for what has been implemented, however
a more complete sip implementation, better technical support, and limited
trouble shooting access will delay my decision to purchase this unit.
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Re: [Asterisk-Users] Mediatrix 1204 sip experience?
See: the ATA section of: http://www.voip-info.org/wiki-VOIP+Phones and: http://www.voip-info.org/wiki-VoIP+Gateways for a list of what is available. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 12:34 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 sip experience? > I'm not sure I understand your english here. I have two x100p's working just fine, > but I've got a couple more pstn lines I'd like to connect up. I probably could > put another one in the system, but I'd rather use a 4-port external gateway that > works well if such a thing exits at a reasonable price. (No, I don't want channel > banks and T1 cards for such a simple environment.) I'm just starting to do the > research on what is actually available. > > > Is it so hard to put X100P as a ethernet device? > > > > I have been trying FXO devices, but gets me luck. > > > > Kannaiyan > > > > - Original Message - > > > > > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip > > FXO > > > 4-port gateway? > > > > > > The archives tend to suggest the box is not very straight forward, and > > possibly > > > lacks some basic pstn interaction features. > > > > > > Thinking about trying one in place of a pair of x100p's (functioning fine > > now). > > > CallerId, etc, supported on this gateway? > > > > > > Rich > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ---End of Original Message- > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Rich Adamson wrote: > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 > Sip FXO 4-port gateway? I have setup a Mediatrix 1204 with there SIP setup. They work! ??? Yes! and No. > The archives tend to suggest the box is not very straight forward, > and possibly lacks some basic pstn interaction features. They are very hard to configure. But on the other hand they have so many settings that they can almost be your PBX system. (Note the almost). > Thinking about trying one in place of a pair of x100p's (functioning > fine now). CallerId, etc, supported on this gateway? as your primary means of getting your lines in. This will slow the inbound calls. It will add 2 rings before it transfer the call to your system. > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
I'm not sure I understand your english here. I have two x100p's working just fine, but I've got a couple more pstn lines I'd like to connect up. I probably could put another one in the system, but I'd rather use a 4-port external gateway that works well if such a thing exits at a reasonable price. (No, I don't want channel banks and T1 cards for such a simple environment.) I'm just starting to do the research on what is actually available. > Is it so hard to put X100P as a ethernet device? > > I have been trying FXO devices, but gets me luck. > > Kannaiyan > > - Original Message - > > > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip > FXO > > 4-port gateway? > > > > The archives tend to suggest the box is not very straight forward, and > possibly > > lacks some basic pstn interaction features. > > > > Thinking about trying one in place of a pair of x100p's (functioning fine > now). > > CallerId, etc, supported on this gateway? > > > > Rich > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mediatrix 1204 sip experience?
UI for switch config allows you to generate scripts for setting if you need them. I found that to be useful. They can be easily configured from remote if you have the UI software. There are features for caller id, but I have not used them yet. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Friday, January 23, 2004 2:40 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Mediatrix 1204 sip experience? Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Rich Adamson wrote: Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? A+ rating for me. What I don't like about 1204: I do not have any windows boxes. The SNMP code they provide to configure the box runs on windows. I had to borrow a lap top just to configure the box. I could not even read the MIB's off the CD on my Sun or linux box. What ever is on that CD really whacks out my Sun and Linux box when I try to read it. What I like about the 1204: After I got them running, I have never had to go back in and do anything to them. Every now and then I will peek at the syslog messages. I have had power outages, many * restarts/reloads and many linux reboots. The 1204's just keep running. I really like that. I has been so long since I configured them, I do not even remember how. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Go for inter-fone products. it can both support sip and h323. - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 2:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Is it so hard to put X100P as a ethernet device? I have been trying FXO devices, but gets me luck. Kannaiyan - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 7:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Works okay but user interface is a little like using RegEdit to program your router. In the version of software the one I have it lack some security features and I am unable to find any DMTF controls - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Sent: Friday, January 23, 2004 2:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? > > Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO > 4-port gateway? > > The archives tend to suggest the box is not very straight forward, and possibly > lacks some basic pstn interaction features. > > Thinking about trying one in place of a pair of x100p's (functioning fine now). > CallerId, etc, supported on this gateway? > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 sip experience?
Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
- Original Message - From: "Ariel Batista" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, November 06, 2003 6:25 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 > > I 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! Thank you for the kind words. > > 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!!--- I am not sure that we should keep going with this conversation on the list since it is only marginally about Asterisk, but a number of other people on the list seem to be interested in the Mediatrix units so... You can use the Scripts functionality of the UME config program to record a configuration and then you can play it back to other units. The chapter on Scripts in the Help -> Documentation menu in the UME gives details on how to do this. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
-- Original Message -- From: "Sean P. Robertson" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] 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. > >Sean I 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!!--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
- Original Message - From: "Bob Knight" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
Although, these directions aren't complete they may help... By default the box restricts outbound calls the settings below should allow any calls to be made. Also, watchout for the you need to dial 1 one pit fall. I was send a 1 from asterisk but somewhere in mediatrix it was getting strip I have to add it back with in the mediatrix. hope this helps MediaTrix 1204 4-Port FXO with Asterisk -Connect a PC that you can change the ip address to 192.168.0.10/24 and the MediaTrix to a Hub/Switch or directly together with a crossover cable. -Set the ip address of the PC to 192.168.0.10/24 -Attempt to ping 192.168.0.1 if successfull skip next step -In the rear of the 1204 their is a factory reset button power down the unit and then despress and power up the unit down for about 10secs. Release and wait about 15secs and then confirm with the ping again -Install the Unit Manager Express software that is provided with the unit and then add the correct MIB -Open Unit Manager EXpress Select SNMP|Preferences|Version|SNMPv1|ok File|open|apaIII_v2_4_10_68.txt (version maybe different this was as of 10/11/2003) Check Automatic Get and type in 192.168.0.1 in the remote ip then click the lightening bolt at the bottom of the screen you should see something indicating SUCCESS -Now you are ready to start configuring the units ip address info ios|org|dod|internet|private|enterprises|mediatrix|experimental|products|Voi pGateway|apaIII|provisionningMib| interfaceGroup interfaceUseDhcp = 0 interfaceStaticGroup interfaceStaticPrimDnsIp = interfaceStaticSecDnsIp = interfaceStaticDefaultRouteIp = interfaceStaticSubnetMaskIp = interfaceStaticLocalIp = gatewayGroup gatewayPort1Group gatewayPort1PermissionMode = 2 gatewayPort1RedirectEnable = 1 gatewayPort1RedirectToAddress = gatewayPort1NetToPstnSourceFilter = gatewayPort2Group gatewayPort1PermissionMode = 2 gatewayPort1RedirectEnable = 1 gatewayPort1RedirectToAddress = gatewayPort1NetToPstnSourceFilter = gatewayPort3Group gatewayPort1PermissionMode = 2 gatewayPort1RedirectEnable = 1 gatewayPort1RedirectToAddress = gatewayPort1NetToPstnSourceFilter = gatewayPort4Group gatewayPort1PermissionMode = 2 gatewayPort1RedirectEnable = 1 gatewayPort1RedirectToAddress = gatewayPort1NetToPstnSourceFilter = ios|org|dod|internet|private|enterprises|mediatrix|experimental|products|Voi pGateway|SignalingProtocols sipMib|sipUAGroup|sipUAServerGroup|sipUAServerStaticGroup sipUAServerStaticRegistrarHost = sipUAServerStaticRegistrarPort = sipUAServerStaticProxyHost = sipUAServerStaticProxyPort = sipUAServerStaticOutboundProxyHost = sipUAServerStaticOutboundProxyPort = - Original Message - From: "Bob Knight" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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. > > > Sean P. Robertson wrote: > > >Ryan Tucker wrote: > > > > > > > >>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 > >> > >> > >> > > > >We are Mediatrix's US distributor and have used them with Asterisk in our > >lab and have had several resellers purchase them to use with Asterisk. They > >seem to work well with Asterisk, but I have to agree that the configuration > >leaves a lot to be desired. Their SIP units use SNMP exclusively and the > >way that their MIB is arranged, it is a little like configuring a Windows PC > >via the registry editor. Thankfully their are only 6 or so settings that > >need to be changed from the default to get it working so once you know where > >everything is, it is not that bad. > > > >One truly embarrassing issue that the current FXO (1204) units have is that > >they are using SNMP v1 and can not be password protected in any way. A new > >version of the firmware will be out in a couple of weeks and will support > >SNMP v3 and will have password protection. Hopefully they will come up with > >a web browser configuration in the future. > > > >Sean > > > > > >___ > >Asterisk-User
Re: [Asterisk-Users] Mediatrix 1204
- Original Message - From: "Ernest W. Lessenger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, November 05, 2003 8:44 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 > > How well does the echo cancel, hangup detect, etc work? We're experiencing > some nasty echos with *, and I need to know whether we can expect better or > worse when we move to a gateway. > The 1204 (FXO) units detect loop drop from the central office just fine and there are no problems with echo cancellation. On another note, we are also a distributor for Digium so I would be very interested in hearing any problems that you are having with their products. If you are having disconnect detection and echo problems please contact me off list and we can try to replicate and correct any hardware issues that you might be experiencing. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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. Sean P. Robertson wrote: Ryan Tucker wrote: 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 We are Mediatrix's US distributor and have used them with Asterisk in our lab and have had several resellers purchase them to use with Asterisk. They seem to work well with Asterisk, but I have to agree that the configuration leaves a lot to be desired. Their SIP units use SNMP exclusively and the way that their MIB is arranged, it is a little like configuring a Windows PC via the registry editor. Thankfully their are only 6 or so settings that need to be changed from the default to get it working so once you know where everything is, it is not that bad. One truly embarrassing issue that the current FXO (1204) units have is that they are using SNMP v1 and can not be password protected in any way. A new version of the firmware will be out in a couple of weeks and will support SNMP v3 and will have password protection. Hopefully they will come up with a web browser configuration in the future. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
At 04:00 PM 11/5/2003, you wrote: We are Mediatrix's US distributor and have used them with Asterisk in our lab and have had several resellers purchase them to use with Asterisk. They seem to work well with Asterisk, but I have to agree that the configuration leaves a lot to be desired. Their SIP units use SNMP exclusively and the way that their MIB is arranged, it is a little like configuring a Windows PC via the registry editor. Thankfully their are only 6 or so settings that need to be changed from the default to get it working so once you know where everything is, it is not that bad. How well does the echo cancel, hangup detect, etc work? We're experiencing some nasty echos with *, and I need to know whether we can expect better or worse when we move to a gateway. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
> > Would 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! > 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/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
-- Original Message -- From: Ryan Tucker <[EMAIL PROTECTED]> >> 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. -rt Would 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 >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
Ryan Tucker wrote: > 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 > We are Mediatrix's US distributor and have used them with Asterisk in our lab and have had several resellers purchase them to use with Asterisk. They seem to work well with Asterisk, but I have to agree that the configuration leaves a lot to be desired. Their SIP units use SNMP exclusively and the way that their MIB is arranged, it is a little like configuring a Windows PC via the registry editor. Thankfully their are only 6 or so settings that need to be changed from the default to get it working so once you know where everything is, it is not that bad. One truly embarrassing issue that the current FXO (1204) units have is that they are using SNMP v1 and can not be password protected in any way. A new version of the firmware will be out in a couple of weeks and will support SNMP v3 and will have password protection. Hopefully they will come up with a web browser configuration in the future. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
On Wed, 5 Nov 2003 14:25:12 -0500, Ariel Batista <[EMAIL PROTECTED]>
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
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[Asterisk-Users] 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 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!
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[Asterisk-Users] 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 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!
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