here is an example of a scenario where codecs cause issues, this example was for CS1000 and avaya red CM:
http://support.avaya.com/public/index?page=content&id=SOLN204147&group=UG_PUBLIC
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try this first:
Configuring SIP Trunks among Avaya Aura Session Manager, Avaya Aura Communicat...
... routes and trunks Administer route list block and distant steering code Administer node SIP parameters Launch NRS Manager Administer service domain Administer SIP gateway...
not sure if you can use the VPN wiz to mess around with the certs, if you are running 5.5.1, you can try this
ftp://ftp.avaya.com/incoming/Up1cku9/tsoweb/11001200/UNIStim_5.5.1/IPCPVA-01.00_00.25.jar.zip
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yup, something about a OLC messages (open logical channel on h323) not working properly, it happened with cs1000 to cs1000 h323 calls as well, i don't think it was limited to cisco/cs1000 only.
sniffer traces showed calls coming off hold did not get the message from the signaling servers. SIP...
i'm not sure if you were looking in to the UniStim ntp, but there is a section there on certificates
not sure what version you are running, but check this doc out
UNIStim Software Release 4.3 for IP Deskphones
http://support.avaya.com/css/P8/documents/100113686...
maybe someone moved the cable (to a different tty)?
or a dipswitch was changed on the SSC (if you have an opt11c)
software change? (Bps changed on TTY)
lots of possibilities....
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in the past i have left the Domain on my laptop to join workgroup, but you have to have admin rights to join the domain again once you are on the Corp network.
to join, you will be asked for the login, so you would log back to your domain with a user name: " corpname\username" as the...
voice mail normally sees the "Original dialed digits" and those don't change even if you forward your call to another user. (as this is one reason why centralized voice mail works)
you can also play around with Second level Forwarding (SFA in CLS of phone in question), but i think that is only...
try putting DANI yes on the phone that is forwarding to Cell.
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page 279 http://downloads.avaya.com/css/P8/documents/100121898
table 75
not sure if that is what will work for you but you can check it out.
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Find a job you love and you'll never work a day in your life. - Confucius
had to turn off the tape drive once, failed eventually, maybe popping the tape out might do it, but if the tape is spinning it might unravel so listen to it and take a chance....
if it keeps stopping at that size even after a reboot, maybe you need to rerun config wiz in standard mode and just...
maybe you need a pickup group, bunch of phones ring, anybody can answer,
but i would go simple with it, and just make an MCR key on a bunch of phones then.....
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maybe DORG is for TIE routes only?
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not on sipline, only as sip endpoints in NRS. downside is you have less control over features etc, but it works.
you if your Sipline install fails, you can add them to the NRS as SIP endpoints (no license required), so long as you have sip trunks, should work ok.
log into the polycom webpage...
what type of system is it (physical, cabinet, chassis, msg, mmg etc)? are the complaints between cabinets? (or IPEs), if this system has MGCs with DSP daughterboards, then it might be the DSPs.....
what type of phones? (IP or TDM?)
what testing has been done so far...
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