Programing authentication ID

[novice2]

Hello all!
My voip provider uses authentication ID along with username and password to authenticate. How would I set the authentication ID on my out bound and inbound trunks.
I have never set up a trunk on PIAF before, but from what I have seen it seems that it will only take username and password.

Any suggestions?

 

[jroper]

Hi

By authentication ID, do you mean that your carrier requires a 4 or 5 digit prefix with each call?

If this is the case, simply add this under trunk in the prefix box.

If not, give us some more information about what your carrier expects and we will endeavor to help.

Joe

 

[novice2]

Hi Joe,
Here is an example of what I am talking about:

SIP Proxy: call.sipprovider.com
User ID: 1234567890(telephone number)
Password: zxwyqkp
Authentication ID: 00dda3625f-1

It would like that, and the provider's server wont register you unless you have all parameters including the authentication ID.

Thanks again, hope that shed some light on it for my sake

 

[novice2]

Sorry...I should have stated that this is what the provider expects me to put inside my ATA device :

SIP Proxy: call.sipprovider.com
User ID: 1234567890(telephone number)
Password: zxwyqkp
Authentication ID: 00dda3625f-1

The prob is that I just dont know to handle the authentication ID part.

 

[jroper]

Hi

OK, I'd suspect the following, but no guarantees, everyone does this differently.
host=call.sipprovider.com
username=00dda3625f-1
secret=zxwyqkp

and put the telephone number in the outbound callerID.

Or you may want something like this:-

host=call.sipprovider.com
username=00dda3625f-1
secret=zxwyqkp
fromuser=00dda3625f-1

in the register string, you could try something like:
00dda3625f-1:[email protected]/1234567890

Try playing with that - also, try a search on the provider with the keyword Asterisk+call.sipprovider.com

If it was difficult, someone may have posted on this before.

Joe

 

[novice2]

Thanks Joe, I will let you know.

 

[novice2]

Thanks again Joe.
I found this:
type=peer
host=162.168.1.111
context=from-trunk-trunk-sip-111-peer
username=106-peer
secret=1234
qualify=yes
on www.cadvision.com/blanchas/Asterisk/pbx106SIPtrunk.html

I then added the authentication ID line and my server registered with the provider.
My problem now is that I am not able to place calls through the outbound trunk. I get a recorded message from my PIAF saying "all circuits are busy now, please try your call again later".

I have ceated the outbound route using the same dial pattern as the outbound trunk.

Your suggestion please. Thank you.

 

[jroper]

Hi

You now need to check where and how the call is not being dropped - e.g. is it trying your service provider, and they are rejecting it, or is it not getting out of FreePBX.

Watch the Asterisk CLI, or check the asterisk log files to see what is happening.

Additionally typing something like set sip debug peer 106-peer may also give you a clue. (Syntax may not be right, I'm still in 1.2 mode)

Joe

 

[novice2]

Hi

You now need to check where and how the call is not being dropped - e.g. is it trying your service provider, and they are rejecting it, or is it not getting out of FreePBX.

Watch the Asterisk CLI, or check the asterisk log files to see what is happening.

Additionally typing something like set sip debug peer 106-peer may also give you a clue. (Syntax may not be right, I'm still in 1.2 mode)

Joe

I am still stuck with the "all cct. busy" recording. At this point I am just .

Here is how things look in my Sip Info.

Sip Registry

Host| Username |Refresh |State |Reg.Time
sip.host.com |telenum |120 |Auth.Sent

Sip Peers

Name/Username |Host |D N ACL |Port Status
1_Out/telenum |x.x.x.x |5060 OK(90ms)
300/300 |x.x.x.x |D N |3355 OK(124ms)

I was trying to include my debug scripts but it wont let me do a copy/paste.

My Trunk setup:

host=sip.Host.com
username=telenum
secret=xxxxxxxxx
auth ID=zzzzzzzzzz
disallow=all
allow=g729&g723&ulaw
type=peer
context=from-trunk
qualify=yes
insecure=very

What do you think Joe? What is it that I am not doing?

 

[novice2]

OK, happy to report that I got this prob solved. The trick was the register string and a proper dial pattern.