SV9100 Can't Forward VE to External # on SIP trunks / Cant Conference 1

[phone_troll]

Howdy,

I've done this many times without using SIP trunks, in virtually the same config, but I am not having any luck. I was first having trouble with forwarding offsite, through my auto-attendant. Then I realized I also could not initiate a Conference call. I just don't know where to look, to fix this. I have many SV9100's with copper trunks and also with PRI's, and have no issues. I suspect there's a setting relating to SIP that I am missing.

As a side note -- I am able to blind transfer 1 external call to another. What I mean is that I can make an external call from a SIP trunk, hit transfer, make a 2nd external call and then hang up successfully. The 2 parties now are talking through 2 of my SIP trunks. I know this isn't quite a conference, but I'm not certain.

Thanks!!

 

[belevedere]

Verify your SIPT carrier is allowing connection of 2 trunks/lines. I believe they can restrict that. Aside from that, verify your feature COS bits that allow CF outside + unsupervised CNF/tk to tk connections.

 

[phone_troll]

Ok, turns out that my inability to conference was a "me" problem. So conference calls do work, but I'm still having trouble forwarding offsite from my auto-attendant. I built VE 2956 to forward offsite, and dialing 2956 works fine from any NEC phone. Once I use option 7 in my Dial Action Table, I am just disconnected when the forward is attempted.

I've tried to add a # at the end of the 10 digit #. That did not work.

I've tried to set the forwarding on Do Not Answer only, instead of ALL calls, in case forwarding was happening too quickly. This did not work.

I'm not sure what to try next.

My work around is to simply have the site record the 10 digit # in the auto-attendant greeting, if it comes down to it. But I'm hoping it's something I'm missing.

I went into the Class of Service for Hold/Transfer, and enabled Call Forward Off-premise. This also didn't work.

This config has worked on many other systems, so I suspect it's a SIP setting on the PBX. Though I've been wrong before...

Thanks!

 

[phone_troll]

Any ideas?

 

[belevedere]

Have you tried a sys spd?
1304, set up a number. You may need to place a # as the last digit for SIPT.
47-13, digit X set to unscreened trf to #2xxxP, feature code + bin number + pause.
NOTE: after setting 13-04, disconnect from the system.

 

[phone_troll]

I haven't. I've never played with the sys spd on 9100s', but it's worth a shot. Thanks.

 

[roldix27]

hi sir what did you did with your conference problem?

 

[phone_troll]

roldix27 -- it wasn't a problem. I just wasn't following how to do a proper conference.

1. While on call, press CONF SOFTKEY followed by dialing the number of party to be added.
2. Press ADD SOFTKEY after 3rd party answers.
3. Press the BEGIN SOFTKEY.

 

[phone_troll]

KISS....Keep It Simple Stupid

So I simply added 2 pauses after my VE, and it worked fine.

I guess SIP life is just that much faster than PRI/copper trunks in this condition. Oh well. It's working.

Thanks belevedere.