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Recent content by rhawkins74

  1. rhawkins74

    SoTel SIP

    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
  2. rhawkins74

    SoTel SIP

    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
  3. rhawkins74

    SIP Trunking HELP!!!

    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?
  4. rhawkins74

    SIP Trunking HELP!!!

    what I did was in the SIP URI I changed the credentials to use credentials user name instead of use credentials auth name
  5. rhawkins74

    SIP Trunking HELP!!!

    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?
  6. rhawkins74

    SIP Trunking HELP!!!

    so I would have to put in a uri for every user on a site?
  7. rhawkins74

    SIP Trunking HELP!!!

    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
  8. rhawkins74

    SIP Trunking HELP!!!

    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...
  9. rhawkins74

    SIP Trunking HELP!!!

    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...
  10. rhawkins74

    SIP Trunking HELP!!!

    yes I see the call coming into the monitor.
  11. rhawkins74

    SIP Trunking HELP!!!

    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...
  12. rhawkins74

    SIP Trunking HELP!!!

    I am assuming the call rejection must have something to do with the SIP URI?
  13. rhawkins74

    SIP Trunking HELP!!!

    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...
  14. rhawkins74

    SIP Trunking HELP!!!

    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...
  15. rhawkins74

    SIP Trunking HELP!!!

    okay, system status is saying the trunk is registered now
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