SOLVED Help, I can't get my Vitelity Trunk to work.

[gcmartin3]

I hope someone can give me a suggestion. I've been fighting this problem for a couple of days already. This is a fresh install of PIAF. (I even installed twice, thinking maybe something went wrong the first time)

The problems I'm having are:
1) On Inbound calls, we get an immediate busy signal
2) On Outbound calls, there is one ring, then it hangs up

I've gone over my trunk and extension settings and they all seem right. Have also gone through the forum and I'm certain I have the same trunk settings recommended.

I've attached four files. Hopefully, someone can see something I've missed. Here's the four files
1) The system status
2) The Trunk Settings
3) The log of the inbound call
4) The log of the outbound call

Please please help......

Thanks much

Best regards
Greg

 

[Mark D. Montgomery II]

I don't have codecs specified (I don't think that should matter?).
I seem to have a number of settings you don't have.

INBOUND:

username=xxxxx
type=friend
secret=yyyyy
nat=yes
insecure=port,invite
host=inbound31.vitelity.net
dtmfmode=auto
context=from-trunk
canreinvite=no

Registration String
xxxxx:[email protected]:5060

OUTBOUND
type=friend
dtmfmode=auto
username=xxxxx
secret=yyyyy
fromuser=xxxxx
trustrpid=yes
sendrpid=yes
context=from-trunk
canreinvite=no
nat=yes
host=outbound.vitelity.net

 

[TwigsUSAN]

Try specifying the codecs. I believe if you don't, then how is the system supposed to know what to use.

 

[Mark D. Montgomery II]

Mine's been working fine without them set, although I suppose setting them wouldn't hurt.

 

[geopeterwc]

I hope someone can give me a suggestion. I've been fighting this problem for a couple of days already.

Have you checked the "Routing Method" on the "Route to Account" for your DID at your Vitelity account? Be sure that the "Routing Method" is set to a "SIP Server" and not "SIP Device".

Pete.

 

[gcmartin3]

Hi. Thanks for the inputs.

I have allow=all on the codecs, and the routing method in Vitelity is set to SIP Server with my static IP.

I am really stuck here....

 

[wardmundy]

Perhaps you should contact Vitelity.

 

[GregKeys]

Show us your inbound routes, outbound routes. In the inbound call you show:
Called SIP/2001
-- Got SIP response 486 "Busy Here" back from xx.xx.xx.xx:37083
-- SIP/2001-0000000a is busy

That "looks" like the pbx tried to connect to 2001 and 2001 replied busy.......
Show us the ext 2001 config too if that is the case, otherwise, tell us who's ip you got that response from..... Your public ip, your private ip, the extensions ip or vitelity's ip

 

[gcmartin3]

Hi. Thanks for the responses. I had also raised a support ticket with Vitelity and they suggested changing the allow=ulaw only.

Extension 2001 is setup with all default values except that NAT is set to YES.

On the endpoint, I've disabled all codecs except G711u. The endpoint is BRIA software running on an iPhone. Very strange though as I have many other extensions setup using Vitelity on this device and the trunk settings had allow=all and they all worked fine.

I made no other changes except for the ulaw setting suggested by Vitelity and now am getting calls through.

Very strange.

Thanks all for your suggestions.