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Yes, I configured a SIP URI on the SIP Line with as local URI '40'. Next in the incoming call route I route '40' to the VM module.

Hello, I'm trying to setup my SOE in combination with Asterisk SIP trunks. Everything is working great except one thing. I'm unable to send incoming calls to a VM Pro module. Is there specific configuration required for this ? I just put in the incoming call route a destination as...

Please note that you cannot share resources between media gateways. There is no common backplane. If you are using no compression (G.711) you can have up to 2 E1/T1 in one G700 unless you add a MM760 which provides you with another 64 DSP resources.

Hmmm, I remember that you could add a phone of type 'h.323' but the problem with applications like netmeeting was that you couldn't enter a pincode which was required by the definity. So yes I you should be able to add a h323 station but registering is something else...

Log off the phone, then you should be able to change the type.

Hello, Since you are talking about VDNs and vector I quess you are using an Avaya CM. If you are refreshing your voice system I also suppose you're at least on CM 3.1 ? In that case you could use vector variables to achieve this and just create diffferent vdn's all pointing to the same vector.

moev99, What does a 'status trunk' say ? The IP address of the Audix is different of the procr one. You should go to the maintenance webpage, select 'configure server' and go thru all the settings. On the page were you define the IP for the S8300 there should be also a field where you put the...

Are the users using digital phones ? Is the G700 grounded ? It depends how the call enters your system and were the recording takes places if you will hear something on the recording or not.

intrigrant, It depends from config to config. Sometimes a CM is even cheaper then an IPO. And to be honest for 10 sites I would prefer real centralised management (one system) instead of being able to reach 10 different systems over the network.

learn2shoot, This is certainly possible. You can access the historical tables by installing the Informix ODBC driver on your pc.

What's the priority of every card and and are they allowed to accept ip phone registrations ?

randyb19, Sorry but you are wrong. You are referring to shuffling (ip-ip direct audio) which under normal circumstances should be on 'Y'. But be carefull, there are reasons you would put this on 'N'. For example if you are using a voice recorder that is sniffing the medpro port instead of using...

Make sure your sipdomain doesn't include a hostname. So 'avaya.com' would be valid, 'support.avaya.com' NOT.

I would advise you to use other building cabling. Try to repatch the extensions reporting problems to other outlets.

jphardy, Did you put the MWI parameter in the settings file ?

Is dadmin enabled in the LSP license ?

Did you configure the proxy selection route pattern on the locations form ?

Indeed, same effect when you install a S8500 or S8700 with the flash drive connected.

Peter, No, the phones support up to 30 gatekeepers (clans + lsps).

Peter, I don't think there is a limit on the number of clans you can put in one region. But the phones are limited to receiving up to 30 alternate gatekeepers when registering. Anyway putting so many clans in one region doesn't give you much granularity to control registration. Why would you...