Best SIP provider for Avaya...

[AMGTEKIII]

We are interested in getting a few SIP trunks to do some testing...Does anyone have any experience with Avaya and SIP??

 

[Midstream]

We are using Global Crossing at two locations. US and UK.

 

[AMGTEKIII]

Thanks for the response...How do you have it set up? Is it working well for you?

 

[Midstream]

We are using SIP trunks in the UK and in the US, will be deploying SIP trunks in Asia within weeks/month. It works reliably, but I understand there are some interactions with conferenceing and faxing.

 

[AMGTEKIII]

Having never used or worked with SIP, what is my best way to work this out?? We want to get some trunks in our lab to start testing and getting familiar with the technology before we roll it out....Any help you could provide would be GREATLY appreciated!!!

 

[mrjedi]

Which country ? We have successfuly used SIP trunks in UK and got DDI numbers for in bound and out bound. Great DR for companies.

 

[AMGTEKIII]

Here in the US, Virginia to be exact.....

 

[sjforcum]

Up in Boston, MA I've got some Paetec SIP trunks running in our lab.

They're testing pretty good right now...

SJF

 

[Midstream]

sjforcum,

Have you tried using Meet Me Conference over those SIP trunks? I've got an installation where the MMC gets multiple prompts. The pass code is valid, the user enters the conference, then gets prompted again, then gets dropped from the conference. If a call comes in on a ISDN trunk to the same vector, no problems at all.

 

[phonegoober]

Rockspop,

are you doing TCP Header Compression on your network?

 

[Midstream]

phonegoober,

Thanks for checking in.

Where would I look to determine that, please?

 

[sjforcum]

Rocks,

I never setup meet me conferening over those trunks (they were in a test lab).

I would check the DTMF setting on the signalling form to start....

SJF

 

[Midstream]

DTMF is set for rtp-payload.

DTMF tones are verified and you get added to the conference successfully. About five seconds later you're prompted for the passcode again, then dropped. If a call comes in a TDM trunk to the same vector, no problem, no drops.

 

[bronco9955]

Check your codec's. Had the same problem a conference bridge from a remote site. Changed to G711 and it flew.

 

[Toni269]

Here is a SIP provider you could try:

http://www.telcentris.com/

Not sure if they are the best because I haven't used them myself but I know people that work there who work hard to earn business and keep customers happy.