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  1. irvjam

    SMGR Certificates

    Currently setting these up at my desk using one IP address and they're not dns - yet... I created csr using openssl had SMGR sign it, then loaded it on two mp114 FXS boxes so far, they both WILL register to SM TLS and can call out fine but they won't accept any incoming calls. traceSM returns...
  2. irvjam

    SMGR Certificates

    Second question. I have to do this for all of these gateways? ie - If I have 20 MP114's in my network I would have to have 20 End Entity's in SMGR, because each gateway has a different CSR, or is there an easier way???
  3. irvjam

    SMGR Certificates

    Thanks kyle555 it works. here's a star
  4. irvjam

    SMGR Certificates

    kyle555, Kyle, Do you have the media pack gateway registering to SM via TLS? I have been spinning my wheels on this for a while now, just heard from Avaya support that only supported configuration port from AudioCodes to Session Manager is TCP. Originally I generated CSR from audiocodes and had...
  5. irvjam

    Avaya 9611 SIP vs 9641 SIP configuration

    I know when I have seen this it has been a TRUST CERTS error in the settings file
  6. irvjam

    Symposium Return to Queue

    One other place to check is Configuration - Skillsets - Call Age Preference should be set to Oldest. If you want priority given to the oldest contact in the system, select Oldest. If you want to give priority to the first contact in the queue, select First In Queue.
  7. irvjam

    Avaya Aura Messaging Back Up

    How does the AAM work with cs1k? any issues outstanding? Do you use session mgr? We are ready to replace our octel 350 with either aam or avst...
  8. irvjam

    CS1K with AVST or Aura messaging???

    We have a network of 12 CS1000's 3 at release 7.5 and 9 at release 5.0. We have a internal network mixture of SIP using Aura system/session manager and t-1's using mcdn. We are replacing our Octel 350 this year (2012. Please reply with any experiences good or bad integrating either AVST call...
  9. irvjam

    one way audio talkslot 30

    That sound right. we have reseated controllers about 4 times now. Seems like whenever we see the BSD messages we need to reseat them. So far we have been running clean for about 3 weeks. Will be upgrading to 7.5 in December so my fingers are crossed for NOW.
  10. irvjam

    one way audio talkslot 30

    Interesting. That's what I was told at first about the patch. Then talking to our vendor thay said it is a bandaide fix for the XMI queue and another permanant patch was developed and currently being tested at another PBX. Now I have 2 loops with 1 way audio and getting BSD202 for these loops...
  11. irvjam

    one way audio talkslot 30

    The issue has something to do with the XMI queue. We have a patch MPLR31095 that fixes this issue. Some other things we noticed from our 7.0 upgrade was: getting several BSD error durng MIDs BSD202 etc. lost our 3902 firmware on disk, XMI005 and AUD571 and AUD572 messages showing up, have to...
  12. irvjam

    one way audio talkslot 30

    thread798-1192586 I am having this exact problem with talkslot 30 on variouse loops and in different network groups. one way audio but always talkslot 30. We are 7.0 pp4 FNF all the latest deplist. Tikimon, if your still out there - What was the fix?
  13. irvjam

    Mobex transfer not working on incomming call

    So i have just setup Mobex. Mid call transfer feature works when i call out using my mobile phone by dialing the MFAC, but cant transfer or conference a incomming call on my mobile phone. Im guessing it's because i am not using a dedicated PRI mobile route and it is basically just a PCA call...
  14. irvjam

    MobEx?

    Looking at setting up MobEx and using the One-X lite with our 7.5 upgrade. That brings up another question, anyone heard about release 8 availability? I am interested in learning about the dedicated route option with mobex. Is this just a separate PRI route to the CO or is there some routing...
  15. irvjam

    sip connection from cs1000 to sipex or aterisk

    I have been trying to connect sip foundry (sipex) to our cs1k. I am able to call out from sipex to cs1k, unable to get calls from cs1k to sipex server. I have noticed others with the same issue using asterisk. I have the sipex server gateway pointing to our node tlan ip address (not the NRS)...
  16. irvjam

    Block CLID on only one PRI

    We have an 81C 23.55 and would like to block sending CLID on our LD ISDN route, but still send the CLID on our local ISDN route. Any ideas? I thought maybe removing COLP from the RCAP but wanted to post this question first.

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