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3300 to Asterisk Trunking 1

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kenny5445

Technical User
Apr 17, 2008
82
GB
Hi, I'm experementing with SIP trunking to a asterisk box.....

I've found the following tutorial,
Which is using trixbox and not vanilla asterisk.

From what I've read you create a ext on the asterisk box (which is registring perfectly) but the asterisk to mitel isnt working I either get a 404 (which means the authentication failed or a time out from asterisk registring.

If anyone has any experiance on this matter help would be appreciated.

I have 1 SIP trunk license for testing (borrowed from a quickconference server).

Thanks

Kenny
 
Which part you need help with?
I have multiple systems working with Asterisk.

 
Slapin, I want to test Mitel to Asterisk, what load are you using for the Asterisk ( sorry to hijack the thread ).
 
As of now Asterisk 1.4.0 and 3300 V9.0 (started with 7.1)

I think it should be fine with latest Asterisk as well.

There is an issue with transfer of SIP calls. I didn't dig too deep, but the best I could get is full media proxy mode. That manss the asterisk makes an outgoing call and forwarding RTP packets. It's OK for LAN environment, but in WAN can cause doubling of RTP traffic.
 
Hi I got it working just by getting the mitel to register as an ext and in the asterisk dial plan route calls to that ext....

I'm thinking about using a YA pro softphone via teleworker for some home agent on a sip dialler (using the trunking).

Anyone see any issue with this? (SIP should stream via the teleworker if i'm correct in my understanding).

THanks
Kenny
 
So, asterisk will work as an autodialer and when someone responds it will call an agent through 3300?

Why don't you connect an agent using IAX or SIP softphone?

Actually your schema will work. YA Pro will be presented on ICP side of the Teleworker gateway and 3300 ICP will give media stream information to Asterisk, so media "connection" will be made from Asterisk to TW.
 
Yeah thats the plan, using YA gives oppertunity for ACD with ccm stats aswell as a nice interface and possibly some collaberation. The quality of telworker phones (used 5330's before) is extremly good via the internet from my my experiance.

The dialler will nail up the end user when they login and conference in the live calls via asterisk...

Cheers

Kenny
 
should be fine. 3300 will take care of translating media information from MiNET (teleworker) to SIP for Asterisk.

 
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