Hi all, 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: to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKa9fa10127;received=98.76.54.32 From: Port 2 <sip:3101@123.45.67.89>;tag=fd593f07870355f To: Port 2 <sip:3101@123.45.67.89>;tag=as52ef97c9 Call-ID: fc3b168b1892aebc2b87f295187ebbea@123.45.67.89 CSeq: 1117525281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3101@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="23e26a38" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 <sip:3102@123.45.67.89> From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER Contact: Port 3 <sip:3102@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf To: Port 3 <sip:3102@123.45.67.89>;tag=as4a4a8cc7 Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3102@123.45.67.89> Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf To: Port 3 <sip:3102@123.45.67.89>;tag=as4a4a8cc7 Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3102@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="48e70a35" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd Content-Length: 0 To: Port 4 <sip:3103@123.45.67.89> From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER Contact: Port 4 <sip:3103@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 To: Port 4 <sip:3103@123.45.67.89>;tag=as4d58c8ce Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3103@123.45.67.89> Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 To: Port 4 <sip:3103@123.45.67.89>;tag=as4d58c8ce Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3103@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="70915041" Content-Length: 0 I'll provide more info, if necessary. Heck, I'll open up my firewall for someone to get into this mediatrix & fiddle with it if they want... Thanks, Jeremy Jones
> 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.
Jeremy, I have tested mediatrix 1104 with * and it worked flawlessly - this however was after I had spent about 3 days trying to register it with *. The trick is to find the realm parameter and change it to ASTERISK. After that there isn't really much to it. I'll try to dig out my snmp walk from the 1104 and post it on the list. Regards, Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of jeremy Sent: Wednesday, May 05, 2004 1:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mediatrix 1104 Hi all, 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: to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKa9fa10127;received=98.76.54.32 From: Port 2 <sip:3101@123.45.67.89>;tag=fd593f07870355f To: Port 2 <sip:3101@123.45.67.89>;tag=as52ef97c9 Call-ID: fc3b168b1892aebc2b87f295187ebbea@123.45.67.89 CSeq: 1117525281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3101@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="23e26a38" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 <sip:3102@123.45.67.89> From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER Contact: Port 3 <sip:3102@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf To: Port 3 <sip:3102@123.45.67.89>;tag=as4a4a8cc7 Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3102@123.45.67.89> Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf To: Port 3 <sip:3102@123.45.67.89>;tag=as4a4a8cc7 Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3102@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="48e70a35" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd Content-Length: 0 To: Port 4 <sip:3103@123.45.67.89> From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER Contact: Port 4 <sip:3103@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 To: Port 4 <sip:3103@123.45.67.89>;tag=as4d58c8ce Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3103@123.45.67.89> Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 <sip:3103@123.45.67.89>;tag=f0384cd93965088 To: Port 4 <sip:3103@123.45.67.89>;tag=as4d58c8ce Call-ID: 05162c90b52053eff39c625479130f96@123.45.67.89 CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3103@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="70915041" Content-Length: 0 I'll provide more info, if necessary. Heck, I'll open up my firewall for someone to get into this mediatrix & fiddle with it if they want... Thanks, Jeremy Jones _______________________________________________ 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
Glad I tried the Audiocodes first! -----Original Message----- From: jeremy [mailto:jeremy@samnjack.com] Sent: Tuesday, May 04, 2004 7:49 PM To: asterisk-users@lists.digium.com Subject: 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 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