similar to: Mediatrix 1204

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

I'm having a problem getting outbound trunking to work using asterisk and an external SIP FXO. 7 digit dialing produces the following output: -- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack -- Called 5925660@mediatrix-1204 -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17 -- Attempting native bridge of SIP/mitel-fe17 and

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

I am attempting to use a Mediatrix 1204 to interface to multizone paging from Asterisk. I have 4 different paging interfaces and want to connect each of those 4 to an FXO port on the Mediatrix. The desired result is to be able to issue some SIP dial string from asterisk, seize the FXO port on the Mediatrix and then have a speech path. I am able to place calls over the Mediatrix when it's

Dear Group, I have been able to configure my Asterisk BOX to receive calls from Mediatrix 1204. I'm having problems sending calls out via my Mediatrix unit. The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends back a Status : 480 Temporarily Unavailable. This is my configuration on Asterisk; exten => _78996.,1,Dial(SIP/${EXTEN:5}@192.168.6.52) exten =>

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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to

I bought a couple of Mediatrix 1204's a few of months back. (Perceived advantages were relatively low overall cost and size per port, and it isn't nearly as vibration sensitive as a PC would be.) Rich Adamson's review from Feb 1 is comprehensive, and the only thing I'd like to add is this: One "feature" of these units that absolutely infuriates me is its behavior for

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

i'm new here and i need help on how where can i get software version 4.0.x of the mediatrix and how can i install it... mediatrix unit im using has a software version of 2.4.9.57. i would like to use H.323 not SIP... please need help asap!... hope to hear from anyone of you soon.. thanks in advance! ------ eder _______________________________ Do you Yahoo!? Win 1 of 4,000 free domain

Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which "port" an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been

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

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

Hi all, Please excuse my newbie status I need help in configuring a mediatrix 1204 PSTN gateway with asterisk. Basically each FXO port is configured with a SIP username and automatic transfer extension, which should transfer incoming calls to an asterisk extension. I created extensions corresponding to the FXO port SIP usernames. Port 1 - SIP username - 21383396 - call forward to - 300

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

We have a Mediatrix 1102 hooked into the network. Both of the attached analog phones and all of their features work, but in the CLI we keep getting "-- Got SIP response 481 "Transaction Does Not Exist" back from XXX.XXX.XXX.XXX " (Where XXX is the IP address of the Mediatrix ) every few minutes. I have changed most of the settings in the sip.conf multiple times and have done

Hi all. I'm evaluating a mediatrix 2fxs 1102. seems great (it has also supervised transfer, that's very needed in office environments and works well). the only I thing I cannot make work is the auth to my asterisk server. If I don't set a password into the mediatrix and *, I can call out, but still the registration goes wrong. using a password, nothing works. I've done some