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...
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???
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...
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.
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...
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.
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...
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...
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?
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...
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...
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)...
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