Re: [Asterisk-Users] SIP gateway question
Hi Rich, Here's how I did it - it's a bit hacky, but it works. It does not require a special extension. In my setup, the 1204 has a static IP of 192.168.1.2; the Asterisk box is 192.168.1.1. For outgoing calls: exten = _NXXNXX,1,Dial,SIP/[EMAIL PROTECTED] Incoming: exten = 192.168.1.1,1,Answer . . (etc. - whatever set of answer routines you use) I've configured the gateway to use Asterisk as its outbound proxy, and also enabled port redirection to point to Asterisk. Problems: provisioning the thing with DHCP is a nightmare. I'm trying to get the DHCP payload to include all of the relevant server addresses, and it works, except for the outbound proxy parameter. The gateway just won't pick it up. It uses everything else perfectly, just not that one thing. I can only conclude it's a bug in the gateway firmware. On long distance calls that start with 0, the gateway always trims the 0 off. I have to add it back in manually with another gateway setting. Highly annoying. Overall, it seems reliable but quirky. Regards, Christian On Saturday 31 January 2004 11:59, Rich Adamson wrote: Hi Bob, The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. I'm trying to figure out how best to bring pstn calls into * using this box, and not sure I'm there yet. Since the box doesn't register with *, I'm using the Redirect method which effectively causes the 1204 to dial x3094. What I'd like to do is simply drop that incoming call into the ivr menu directly. Any thoughts on how best to do 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 ___ 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] SIP gateway question
Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to register like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm using canreinvite=no to forcably keep * in the middle for now. Questions: 1. The 1204 answers incoming pstn calls correctly, cycles through the invite/ trying/ringing (I have * config'ed to simply ring an internal sip phone for testing purposes), and I answer the call just fine from the sip phone. When I hang up the sip phone, * sends a Bye and the 1204 says OK. The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? 2. The 1204 seems to be set to a 30 millisecond sampling rate while all other sip phones, etc, are set to 20. Anyone have any thoughts as to whether that would cause a problem later, or should I change that to 20 milliseconds for consistency? 3. Has anyone played with this box and found any unusual problems, weird config's, etc? The box is essentially in a test/eval mode, anticipating using it to replace a couple of x100p's. 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] SIP gateway question
Rich Adamson wrote: The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. -- 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] SIP gateway question
Hi Bob, The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. I'm trying to figure out how best to bring pstn calls into * using this box, and not sure I'm there yet. Since the box doesn't register with *, I'm using the Redirect method which effectively causes the 1204 to dial x3094. What I'd like to do is simply drop that incoming call into the ivr menu directly. Any thoughts on how best to do 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
