DEAL New Kid on the Block: V1VoIP

qtlnx

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Dear @wardmundy ! I've read your instructions several times but could not figure out how apply them, particularly because FreePBX SIP trunk wiki hasn't been updated since 2014 - screenshots don't match current.
"A picture worth a thousand words". That is why I posted my screenshots in expectation that you can do the same, instead you keep repeating
create two separate trunks
over and over again. Why not three separate trunks? There is sip3.v1voip.com also!
 

wardmundy

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The settings already were in my post that you quoted. You just name the trunks v1voip1 thru 3 (now), and enter the settings provided in the PEER DETAILS. Everything else can be left blank. You do have to enter your IP address in your trunk setup on the v1voip side.

Code:
host=sip.v1voip.com
type=friend
insecure=port,invite
qualify=yes
disallow=all
allow=ulaw
nat=no
dtmfmode=rfc2833
canreinvite=no
context=from-trunk
For second trunk, host=sip2.v1voip.com. For third trunk, sip3.v1voip.com.
 

qtlnx

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and enter the settings provided in the PEER DETAILS
settings in PEER DETAILS are responsible for outgoing calls which work just fine for the screenshots provided.
I am talking about incoming calls.
You do have to enter your IP address in your trunk setup on the v1voip side.
I can't enter IP address because my IP is dynamic. The registration line is designed to provide my current IP address to v1voip. I am getting incoming INVITE but PBX responds with 401.

Again, I have no problems configuring PJSIP trunk with v1voip but would like to be able to use SIP trunk as well.
 

Eliad

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V1VOIP requires a fixed IP address for the user. Other providers require an id and a password, some providers can go both ways, different designs by the VOIP provider.
 

qtlnx

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V1VOIP requires a fixed IP address for the user. Other providers require an id and a password, some providers can go both ways, different designs by the VOIP provider.
I disagree since it works for me just fine and I never explicitly provided my IP address to V1VoIP. I can send/receive calls from any IP without changing anything on V1VoIP side.
 

krzykat

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I disagree since it works for me just fine and I never explicitly provided my IP address to V1VoIP. I can send/receive calls from any IP without changing anything on V1VoIP side.
What people are telling you is that in order to mitigate fraud, many carriers want to make Damned sure you are you. To that end, you provide them with your IP address that will be sending out traffic. They essentially use a whitelist like many of us do to insure that there is no fraudulent use. You, the client, by entering your IP address of your equipment are authorizing any traffic that comes from that IP as being from you and you will be billed for it. Going straight on user/pw just opens things up for fraud and abuse, and trust me, you don't want to be that person that gets hit with $100K phone bill.
 

sortons

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There are two ways you can configure the inbound calls; both start on your PBX side with Ward's template. On the V1VoIP portal side on the "DIDs' -> 'My DIDs' page click on the Forward icon on any DID (it's the right-most icon - mouse-over shows 'Forwarding'.)

1. IP based: Chose 'Forward to 'IP Address/PBX' and in the field below type the IP of your PBX and make the T/O 25-30 seconds. The way my DIDs work is (on the same page at the bottom under 'SIP Type') US DIDs are 11 digits and Canadian ones 10 (that's the only way I get incoming properly routed to the DIDs). No need to alter any other information on your trunk's side.

2. Auth based: On the same page as above, chose 'Register Device/Trunk' and there will be two fields added below with your 'Username' and 'Password'. Copy that information in the following format on your trunk's 'SIP Settings' -> 'Incoming' -> 'Register string':
'Username':'Password'@sip.v1voip.com/'Username' - replace info as described above and remove the quotes. The T/O and 'SIP Type' comments from above apply here too. The rest of the fields on that page remain empty.
NOTE: this approach works with ATA devices too.

If you have many DIDs you can pre-configure the option (IP based or Auth based) on the 'Pre-configuration' page under your name (right-most pull-down on the top of the page) on their portal.

Also, note @krzykat's comment - while I agree that both those options may work without defining the IPs under 'Termination' -> 'Endpoint Request' it's worth spending the additional seconds to put in those IPs even if dynamic (just change them as required.)

Although the portal is quite basic, the price / options / voip quality ratio is quite attractive.
 
Last edited:

wardmundy

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Thanks for the update. auth-based authentication has been added since we signed up.
 

qtlnx

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2. Auth based: On the same page as above, chose 'Register Device/Trunk' and there will be two fields added below with your 'Username' and 'Password'. Copy that information in the following format on your trunk's 'SIP Settings' -> 'Incoming' -> 'Register string':
'Username':'Password'@sip.v1voip.com/'Username' - replace info as described above and remove the quotes. The T/O and 'SIP Type' comments from above apply here too. The rest of the fields on that page remain empty.
NOTE: this approach works with ATA devices too.
Thanks for clarification for others. I've been using option 2 since I first come across v1voip. That is why I didn't understand why @wardmundy kept insisting on option 1.
However I am still puzzled with required User Details contents here:

Could anyone just provide a clear example? Do I really need /1nnnnnnnnnn at the end of register string?
 

wardmundy

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Thanks for clarification for others. I've been using option 2 since I first come across v1voip. That is why I didn't understand why @wardmundy kept insisting on option 1.
However I am still puzzled with required User Details contents here:

Could anyone just provide a clear example? Do I really need /1nnnnnnnnnn at the end of register string?
I posted the User Detail settings above. You need two (now apparently three) separate trunks as noted in the post. The /1nnnn entry on the end of the register string tells FreePBX which of your outbound routes to send incoming calls to. So, if you put /12345 on the end of the register string, then you should have an outbound route defined for 12345.
 

wardmundy

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Leave User Details blank. Sorry for the confusion.
 

sortons

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Two things to check:

1. Are you using the same username and password on the register string of the SIP trunk as the ones in the functional PJSIP trunk?
2. Have you disabled the PJSIP trunk before enabling the SIP one?
 

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