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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 __________________________________________________________ Find a job you love and you'll never work a day...

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 __________________________________________________________ Find a job you love and you'll...

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.... __________________________________________________________ Find a job you love and you'll never work a day in...

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. __________________________________________________________ Find a job you love and you'll never work a day in your life. - Confucius

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. __________________________________________________________ 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..... __________________________________________________________ Find a job you love and you'll never work a day in your life. - Confucius

maybe DORG is for TIE routes only? __________________________________________________________ Find a job you love and you'll never work a day in your life. - Confucius

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...

i think the Berkeke communicates via SIP trunks (guessing). so you would need to point it at a NODE IP with Sip trunks (of the CS1K). then you would need to setup your NRS as well for the dialplan to work. __________________________________________________________ Find a job you love and you'll...

if you want to double check, take a look at page 307 on this doc. http://downloads.avaya.com/css/P8/documents/100120098 __________________________________________________________ Find a job you love and you'll never work a day in your life. - Confucius

on the ACD did you chose IVR yes and ALOG yes? What is the status of the agents? (on callpilot, pbx, AACC). can you log in to Voice Item maint and listen to your recording for that callpilot App? if you can include some printouts, maybe someone else can catch whatever is preventing it from...

the reason is you are in a sig sever from the looks of the capture. you are using the command for the PBX part (vxworks). maybe you just have the wrong IP address (assuming you are trying to get into the VxWorks pbx), look for the active ELAN ip of the CALL SERVER...

i don't believe so, the SIP registrar part changed on 6.0 so it was a real SIP proxy, and for the polycom to work properly, it needed the 6.0 on 5.5 you will most likely get redirect errors since it was not a true sip proxy on 5.5 and below. the call presentation and setup is handled by the NRS...