Re: [Asterisk-Users] mediatrix 1104
I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway. There's no printed documentation shipped with the unit, but I have a piece of software for windows that shipped with a different model (which I haven't tried configuring yet), that uses snmp to set misc variables (ip settings, sip stuff, etc.). Fairly baroque interface pretty slim on help... Basically, I'm wondering if anyone's ever configured one of these things for use with *, if anyone could share any tips with me... Doesn't seem like I'm getting it to register w/* -- I thought I'd been setting the proxy username/password in this thing, but I keep getting this with sip debug: Seems all of the Mediatrix stuff is configured through snmp only. Finding and changing the parameters is a royal pain, however others have posted to the list using that same model. I would stay away from their fxo model however. After many hours of working with a reseller, ended up having to send it back. Mediatrix's gameplan seems to be oriented towards selling the fxs and fxo boxes in pairs as a form of toll bypass. They really aren't interested in standards and making their products work with *, etc. ___ 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 1104
Rich et alia, Seems all of the Mediatrix stuff is configured through snmp only. Finding and changing the parameters is a royal pain, Yer tellin' me! however others have posted to the list using that same model. Really? I wasn't able to come up with anything googling, other than someone else asking how to configure the things... Please, throw up a link if you see something I don't. I would stay away from their fxo model however. After many hours of working with a reseller, ended up having to send it back. I'm on the verge with this one. Mediatrix's gameplan seems to be oriented towards selling the fxs and fxo boxes in pairs as a form of toll bypass. They really aren't interested in standards and making their products work with *, etc. But one would think it'd be fairly simple to at least do a straightforward sip proxy registration, no? Anyhoo -- I'll beat on it now then for a couple days post results if anyone's interested. In the meantime, my offer to open up access to anyone who'd like to take a stab at it is still on the table. Thanks, Jeremy ___ 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 1104 Configuration
Mark, Looking for 201 in default Mar 7 14:31:37 NOTICE[65541]: pbx.c:1212 pbx_extension_helper: Cannot find extension context 'default' In my extensions.conf I have the following defined. [default] include = sip exten = 333,1,VoicemailMain exten = 333,2,Hangup [conference] exten = 801,1,Meetme,5010 exten = 802,1,Meetme,5020 [sip] include = conference include = default exten = 201,1,Dial(SIP/201,20,tr) exten = 201,2,Voicemail,u201 exten = 201,102,Voicemail,b201 Suggest playing around with different context names structures From above: [default]: include sip; [sip]: include default appears to be rather circular, and I wouldn't even try to guess at how * is going to treat that. 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 1104 Configuration
Hi Rich You never did tell us what the problem is that you're trying to solve, or what you've done to help identify whatever the problem happens to be. Thanks for responding. What I am doing is I have 2 x Mediatrix 1104 boxes which I will be plugging analog phones into. I found the initial problem it was a missing username = 246 in the sip.conf I now have it set to [201] type=friend username=201 secret=201 host=dynamic dmtfmode=inband nat=no context=sip [246] type=friend username=246 secret=246 host=dynamic dmtfmode=inband nat=no context=sip And the mediatrix registers properly. However now I get a new problem :-) When I dial 201 from 246 I get the following error Looking for 201 in default Mar 7 14:31:37 NOTICE[65541]: pbx.c:1212 pbx_extension_helper: Cannot find extension context 'default' In my extensions.conf I have the following defined. [default] include = sip exten = 333,1,VoicemailMain exten = 333,2,Hangup [conference] exten = 801,1,Meetme,5010 exten = 802,1,Meetme,5020 [sip] include = conference include = default exten = 201,1,Dial(SIP/201,20,tr) exten = 201,2,Voicemail,u201 exten = 201,102,Voicemail,b201 exten = 333,1,Ringing exten = 333,2,Wait(2) exten = 333,3,VoicemailMain exten = 202,1,Dial(SIP/202,20,tr) exten = 202,2,Voicemail,u202 exten = 202,102,VoiceMail,b202 exten = 225,1,Dial(SIP/225,20,tr) exten = 225,2,VoiceMail,u225 exten = 225,102,VoiceMail,b225 exten = 226,1,Dial(SIP/226,20,tr) exten = 226,2,VoiceMail,u226 exten = 226,102,VoiceMail,b226 exten = 227,1,Dial(SIP/227,20,tr) exten = 227,2,VoiceMail,u227 exten = 227,102,VoiceMail,b227 exten = 229,1,Dial(SIP/229,20,tr) exten = 229,2,VoiceMail,u229 exten = 229,102,VoiceMail,b229 exten = 232,1,Dial(SIP/232,20,tr) exten = 232,2,VoiceMail,u232 exten = 232,102,VoiceMail,b232 exten = 243,1,Dial(SIP/243,20,tr) exten = 243,2,VoiceMail,u243 exten = 243,102,VoiceMail,b243 exten = 246,1,Dial(SIP/246,20,tr) exten = 246,2,VoiceMail,u246 exten = 246,102,VoiceMail,b246 I guess I have missed something really obvious but cannot for the life of me figure it out. Any pointers would be greatully recieved. Regards Mark ___ 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 1104 Configuration
Hi All However now I get a new problem :-) Just to let you know it was my incompetance that caused the problem and I now have it all working as expected :-) Regards Mark ___ 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 1104 Configuration
Mark, You never did tell us what the problem is that you're trying to solve, or what you've done to help identify whatever the problem happens to be. I don't have a 1104 but have many hours invested in trying to get the 1204 to function with asterisk. Depending upon exactly what you're trying to accomplish, you might want to consider turning on the 1104's syslog debug option and forward those events to a syslog server. There is a fair amount of detail contained in those debug logs. Hi All I am having a problem with the configuration of a Mediatrix 1104. In the mediatrix i have in the SIP section Registrar:192.168.62.200 Proxy:192.168.62.200 Under the sip user agent I have Username = 246 Friendly name = 246 The i configured the authentication for realm asterisk user 246 password 246 I can ping the mediatrix gateway from the asterisk machine and both are on an internal network with no firewalls inbetween. In asterisk I have [246] type=friend host=dynamic nat=yes ;defaultip=192.168.0.1 username=246 secret=246 mailbox=246 canreinvite=no context=sip callerid=Test 246 If I enable sip debug in asterisk I dont see any authentication failures or anything :-( Any suggestions or sample configs anyone could send? Thanks in advance Mark ___ 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
