GO HERE V1VOIP Help with Settings

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
New to this forum, I posted a message earlier and made quick successive edits to correct mistakes. That probably lead my post to go into review, so it is not visible on the board (yet). Unfortunately time is running and the day is drawing to a close. If there is a moderator, I would appreciate it if my post can be released so that maybe someone can help.
Otherwise, a quick run. I'm trying to get a SIP URI trunk setup on IncrediblePBX (v12.0) / Asterisk (v13.10). I followed Ward Mundy's instructions, which do not clearly state how the outbound trunk is supposed to be set.
Any help will be appreciated.
Sorry if I broke any forum rules earlier.
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
Thank you krzykat. I'll try your config in a moment. I was under the impression that v1voip required a custom trunk with no Peer Details, but a simple SIP/[email protected] I stand corrected!
The good news is, earlier I was able to get the incoming sip uri to work.
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
No joy! I created the trunk exactly the same. I set an outbound route to use it. I immediately get a "All circuits are busy now" message.
Earlier I turned on "Allow anonymous incoming SIP calls" in Asterisk SIP settings and "Allow SIP guests". I guess they only impact incoming sip uri calls anyways right? And my incoming SIP URIs from v1voip work well now.
Also I am using the chan_sip driver. Does it matter?
As far as the PBX goes it is behind pfsense but it can see term.v1voip.com as it can ping it.
 

atsak

Guru
Joined
Sep 7, 2009
Messages
1,822
Reaction score
186
Do you have your IP address set correctly as a permitted termination endpoint? What does the log say (/var/spool/asterisk/full)
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
Yes, it is properly set, and currently used for incoming calls from v1voip. My outgoing calls do not leave my PBX for some reason.
Edit: I meant to say the same IP address is entered in both DID and termination settings on v1voip's dashboard. Since, I am getting proper DID routing by SIP URI, I am assuming that my IP is properly validated by their system. I actually spoke with their support people earlier. So, IP is not the issue.
 
Last edited:

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
The jury is still out for the logs on /var/log/asterisk/full. The file is huge and I can't figure out much. What should I be looking for?
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
Just curious if anyone out there is using SIP URI termination on on older FreePBX/Asterisk (v12.0/v13.10) with the chan_sip driver. I am wondering if this is not some sort of software bug.

@krzykat: What software version are you running?
 

krzykat

Guru
Joined
Aug 2, 2008
Messages
1,579
Reaction score
427
Location
South Florida
Just curious if anyone out there is using SIP URI termination on on older FreePBX/Asterisk (v12.0/v13.10) with the chan_sip driver. I am wondering if this is not some sort of software bug.

@krzykat: What software version are you running?
Pretty much the same. This was Asterisk 13.19.2 / Incredible 12.0
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
@wardmundy: Currently, I have the following as I was told by v1voip to use term.v1voip.com. To make sure that there was no DNS resolution issue I put in the IP address.

Under Peer Details:

type=friend
qualify=yes
nat=no
insecure=port,invite
host=207.239.159.164
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw

I keep getting "All circuits are busy now" as soon as I dial a number using it.
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
@krzykat : On v1voip's termination dashboard, are you using a "Tag" or a "Prefix"? Or do you have them both blank?
 

Halea

New Member
Joined
Aug 12, 2016
Messages
14
Reaction score
1
Problem has been solved. It turned out that it was not pbx related but an obscure 1:1 NAT issue.
Thanks for all the guidance and help you provided.
 

Members online

PIAF 5 - Powered by 3CX

Forum statistics

Threads
22,450
Messages
138,038
Members
14,613
Latest member
roshan2019