I will have a chance to test VOIP Innovations SIP TRUNK. SO far I get a response from seller: The welcome letter has our IP address on it to conect to get wholesale service. Once that is established you can go into the back office and order DID. Once ordered it is provisioned at that time and you will be able to send traffic.

This is the answer for my question: Could you provide Simple SIP Trunk configuration?


DOn't understand that but if anyone has already SIP trunk configured I would appreciate general info about SIP TRUNK Configuration.

I can't find anywhere else about general trunk config at least some sample..... Also their page www does not provide any info.

Thank You,

Trunkname: VoIp Inno

Outgoing:
type=peer
trustrpid=yes
dendrpid=yes
host= (the IP address they give you)
disallow=all
context=from-trunk
canreinvite=no
allow-ulaw&g711
insecure=very


Internal:

User Contex (phone number they give you, like 19365551212)

canreinvite=no
context=from-trunk
fromuser=(number they give you to receive calls on like 1936XXXXXXX)
qualify=no
type=user


Register String
(complete phone nember & IP address like 19365551212@555.555.555.555)

This works on mine currently so should work on yours.


Then you will need to set up incoming and outgoing dial route.

In addition goto "documentation" on red bar at top of page, select it and then select
part way down on left "Elastix without Tears". Will help you as a newbee if no prior knowledge and will help you set up in and out dial rules.

Feel free to ask if you need addition info.

Good luck

Thank You Amphibian, I will try it tomorrow.

Cheers,

It does not work, I have request sent and Sip peers Unmonitored.

MST

Well, first question is do you have any other VoIp trunks working or is this the only one?
Second, are you behind a firewall, like using it with your home dsl or cable and are you in the U S?

Cheers

IT works like a charm. I have inbound and outbound configured, and I am able to get and make calls. So, everything works. However, only my supprised arose to that SIP registry is 120 REQUEST SENT and SIP peers is UNMONITORED.

regrading your questions, yes there are other trunks configured and Elastix is behind CISCO ASA firewall.

MST

Glad to hear it works.

As to the 120 request and such, I would check to see that all ports required are open and forwarded correctly.
If using firewall on server make sure that it's passing the required traffic.

VOIP Innovation, I have found, is a great VOIP company to do business with. The have a great Tech support area, will help (for free) when they can and will work to get probs taken care of quickly. Paul Frey is a good Tech contact to have on your side, educated and great personality.

Prosper well, make lots of money and don't forget the guys at Elastix, they're always here to help.



William

Thank You William for your help and have a good one.

MST

It looks like DTMF tones are not recognized if we call outside number like 800 and after we press whatever is in options we have a message "The number is not recognized" ....

I place dtmfmode=rfc2833 in trunk config under peer but still no luck .... simply outbound DTMF is not recognized.

Could you please advice?

Thank You,

Looks like dtmf=inband resolved that problem.

THX