Problems with vitelity & pbiaf

[manderso]

Solved-Problems with Vitelity, now other questions

*This is solved, but I have questions on post 15.
I screwed up my previous post, so this one is probably going to be short. I've learned (hopefully) a lot on this forum and nerdvittles.

I'm able to make phone calls, but unable to receive. I get a busy tone when trying. I get the standard email from vitelity, saying the server isn't registered.

According to Vitelity: Your current default routing method is set to a SIP server or SIP extension gateway device.

I've checked through the forums, and checked my firewall to verify that my ports are forwarded correctly, 5004-5084 (udp) are forwarded to my pbiaf machine, as well as 10000-20000(udp).

I'm quite stuck, as this has worked previously. The only thing that I changed between when it worked and when it didn't was changing the dialout pattern attempting to make the dialout process faster. Please help. Thanks.

The inbound settings are:

type=friend
dtmfmode=auto
username=bmanderso
secret=
context=from-trunk
insecure=very
canreinvite=no
host=inbound22.vitelity.net:5060

The register string has my [email protected].

Edit: Fixed in post 12.

 

[ronw]

I

The register string has my [email protected].

The register string should be:

bmanderso:[email protected]

Use the Vitelity Communications Setup Wizard
from the support link. It gives the trunk configuration and registration string.

 

[manderso]

Thanks, I'll give that a try when I get home.

 

[manderso]

The register string should be:

bmanderso:[email protected]

Use the Vitelity Communications Setup Wizard
from the support link. It gives the trunk configuration and registration string.

So I change it to that register string, and on calling it, get "The number you have dialed is not in service."

 

[phinphan]

Do you have an inbound route for the vitelity DID or an inbound route for all dids?

 

[manderso]

The vitelity is the only inbound route. Is that what you're asking?

 

[phinphan]

In Freepbx, Inbound Routes do you have an inbound route configured for the vitelity DID? The message you are getting says to me that you are registered but when the call is hitting your pbx it has no route and therefore you are getting the message from Allison saying The number you have dialed is not in service. Your inbound route should allow all did and all cid's and have a destination, i.e. your extension or voicemeail or something.

 

[manderso]

In Freepbx, Inbound Routes do you have an inbound route configured for the vitelity DID? The message you are getting says to me that you are registered but when the call is hitting your pbx it has no route and therefore you are getting the message from Allison saying The number you have dialed is not in service. Your inbound route should allow all did and all cid's and have a destination, i.e. your extension or voicemeail or something.

I've got the two trunks configured, I have the outbound route configured (and can make calls from it, albeit poor quality), but I have no inbound routes configured. There aren't instructions on the vitelity website for freepbx for that. Just "Step 1: Create an Inbound Trunk called 'vitel-inbound'
Step 2: Create an Outbound Trunk called 'vitel-outbound'
Step 3: Goto the outbound routes page and add vitel-outbound to the list of routes."
So I added an everything inbound route, and assigned it my telephone number, but couldn't find a place to assign a trunk. Of course, I get the same error when I call it.
Thanks for your help so far. I hope I'm not being too obtuse.

 

[JohnBrady]

First: In the Vitelity Portal - you need to set the destination for the DID - either to your IP address or your user account. Choose The DID> dropdown the action tab to Routing> choose the appropriate routing of the inbound calls.

Next: You need to have an Inbound route set up in freepbx that describes what you would like the system to do when it receives this call - (ring and extension - maybe an automated attendant, etc.)

Remember the inbound call hits vitelity and it needs to know where to send it > then when it is sent to your pbx, your pbx needs to know what you would like it to do then.

Hope that helps some. - the directions are in the support tab on the Vitelity portal site. (you need to log into vitelity.com to get to the portal)

 

[phinphan]

It sounds like your registering okay and failing when the call hits your pbx. One way to check is to configure a failover line at vitelity that points to your cell phone or another number. Call the vitelity number and if it rings the failover line, your pbx is not registered. If it does not ring the failover line, your pbx is registered with vitelity.

To configure the inbound routing and in general I suggest you read the following:

http://dumbme.mbit.com.au/piaf/piaf_without_tears.pdf

Section 8.1 deals with inbound routing.

 

[rossiv]

This may help. I had the same problem, except with a Vonage SoftPhone account. Try allowing anonymous inbound SIP calls under Setup > General Settings. That was the only thing that would work for me. Hope it helps!

 

[manderso]

It sounds like your registering okay and failing when the call hits your pbx. One way to check is to configure a failover line at vitelity that points to your cell phone or another number. Call the vitelity number and if it rings the failover line, your pbx is not registered. If it does not ring the failover line, your pbx is registered with vitelity.

To configure the inbound routing and in general I suggest you read the following:

http://dumbme.mbit.com.au/piaf/piaf_without_tears.pdf

Section 8.1 deals with inbound routing.

Section 8.1 did indeed help me fix the problem. I had my own number in the did number field. Removing that solved the problem. Thanks very much!

 

[manderso]

Another question. How do I make the call time less? There is a 13 second span between the last digit dialed and the first ring. There has to be a way to shorten that, right?
Thanks.

Edit: Seeing as I'm using the Pap2T ATA as the extension, I searched for some support for it and found this. He's saying to disable the * codes for them internally. Does he mean this?

By manderso at 2009-09-19 If so, how do I disable them, delete them? Will this help the dial out time?

 

[parker]

Another question. How do I make the call time less? There is a 13 second span between the last digit dialed and the first ring. There has to be a way to shorten that, right?
Thanks.

Edit: Seeing as I'm using the Pap2T ATA as the extension, I searched for some support for it and found this. He's saying to disable the * codes for them internally. Does he mean this?

By manderso at 2009-09-19 If so, how do I disable them, delete them? Will this help the dial out time?

Deleting or disabling the "*" codes is useful for letting asterisk (running on the server) handle those functions, as opposed to the software in the PAP2/T handle them. The latter is almost always better than the former, although it doesn't seem related to your question.

Is the "delay" happening on dialing out or in? If the former, try pressing "#" after entering all your digits. If the latter, it could be due to asterisk eating the first ring so as to capture the callerid of the inbound caller on an analog/zap/pstn line (in the US this appears between the first and second rings).

 

[manderso]

Deleting or disabling the "*" codes is useful for letting asterisk (running on the server) handle those functions, as opposed to the software in the PAP2/T handle them. The latter is almost always better than the former, although it doesn't seem related to your question.

Is the "delay" happening on dialing out or in? If the former, try pressing "#" after entering all your digits. If the latter, it could be due to asterisk eating the first ring so as to capture the callerid of the inbound caller on an analog/zap/pstn line (in the US this appears between the first and second rings).

This is happening on the dial out. I'll try the #. Thanks.

 

[manderso]

This is happening on the dial out. I'll try the #. Thanks.

That did the trick! Thanks a lot for all the help.

More questions, if I may.
On home voip plans, who do you (forum members reading this post) go through (provider wise), do you have multiple providers for redundancy, and do you pay for the unlimited plans, or per minute?

Also, regarding entire house setup. I currently go through Charter for our phone system, and I think they use a voip setup, as they have (I think) all our phone jacks hooked into one RJ-11 going into the big box inside the house that seems to shell out the telephone, cable internet and cable TV. I should be able to just plug that RJ-11 into my Linksys PAP2T without having to do anything else, right? I read somewhere that the line needed to be disconnected from the provider, and I'm not certain where to do that.

Thanks again for the help.

 

[manderso]

Anyone? Bueller?