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M8trx, I have looked at the application notes in the knowledge base. That was for r4, and quite a few things have changed in manager since then

Does anyone have any experience setting up a SIP Trunk with SoTel? They are giving us a test line and wanted us to market to our customers, but I am unsure of setting up the trunk on a 500v2 r8

they are listed as the same on mine, but for some reason it did the trick. one last question for you though, is it normal for the SIP trunk to not have a dial tone?

what I did was in the SIP URI I changed the credentials to use credentials user name instead of use credentials auth name

okay, I have now gotten it to make and recieve calls. Since I have 2 trunks should I set the max calls at 2 for the SIP URI or should I create a second channel?

so I would have to put in a uri for every user on a site?

tlpeter.. well, since it is registering, I would assume it is. I am now able to receive calls on the sip trunks, just not make them

okay, I set PAI to None and still recieving the same thing 134568mS Sip: License, Valid 1, Available 10, Consumed 0 134622mS Sip: CMMediaSTUNFilter::callback_received addr f5a65380 (rtp f5a65494 rtcp f5a65380) rec rtp 0 rtcp 0 video rtp 1 video rtcp 1 142530mS PRN...

This is what I get in the monitor when trying to place a call: 665807mS Sip: License, Valid 1, Available 10, Consumed 0 665861mS Sip: CMMediaSTUNFilter::callback_received addr f5a65494 (rtp f5a65494 rtcp f5a65380) rec rtp 0 rtcp 0 video rtp 1 video rtcp 1 667859mS Sip...

yes I see the call coming into the monitor.

Channel 1 Via 10.0.1.190 Local URI: Use Credentials User Name Contact:Use Credentials User Name Display Name: Use Credentials User Name PAI: Use Credentials User Name Registration set to SIP Credential Incoming Group: 0 Outgoing Group: 2 Max Calls per Channel: 2 ARS Route Id 51 Route Name SIP...

I am assuming the call rejection must have something to do with the SIP URI?

Oh I understand fully about the full Monty, I figure once I get this up and running properly, then I can save the config and start playing around on it, and work on setting it up in a full environment. I set up the ars and the shortcode and I get a call rejected when trying to use them. also...

NLPeterK... Let me say thank you so far for your help on this. As for setting up incoming and outgoing, we are wanting to keep it very simple. Like I said, this is one of our demo kits, so there is no one being assigned DID's or anything of that nature. As of right now, we have it set up...

okay, system status is saying the trunk is registered now

Also I had ran stun with the server you said

100385mS SIP Call Tx: 17 INVITE sip:N@sbc.voipdnsservers.com SIP/2.0 Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK237242247d51a5696978ed2356642473 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d...

1246895mS SIP Reg/Opt Tx: 17 OPTIONS sip:Unknown@sbc.voipdnsservers.com SIP/2.0 Via: SIP/2.0/UDP 10.0.1.190:5060;rport;branch=z9hG4bKbb5d9651b59e014b779f0b2eb5da6326 From: <sip:Unknown@sbc.voipdnsservers.com>;tag=91427dc0e5bf9002...

No I had not run STUN

346895mS SIP Reg/Opt Tx: 17 OPTIONS sip:Unknown@sbc.voipdnsservers.com SIP/2.0 Via: SIP/2.0/UDP 10.0.1.190:5060;rport;branch=z9hG4bK6fc9c5e3db8aa06acd2844a6659a13b2 From: <sip:Unknown@sbc.voipdnsservers.com>;tag=9c6137c9f6083e2f...