internet forwarding Not Working?

[kieranmullen]

I am using freepbx and asterisk.

Peer details
authuser=MYNUM
canreinvite=yes
context=global
dtmf=inband
dtmfmode=inband
fromdomain=sanfrancisco-1a.vtnoc.net
fromuser=MYNUM
host=sanfrancisco-1a.vtnoc.net
insecure=very
nat=no
qualify=yes
secret=PASS
type=peer
username=MYNUM


Incomming Settings - None

Reg string MYNUM:PASS@sanfrancisco-1a.vtnoc.net/MYNUM

I have setup a incomming route...

But regardless of whether or not incomming is setup correctly... internet forwarding should work...

I am using the same settings as my ipkall.com number user@domain.com and it is not working.

Just wondering if anyone else is having issues?

KM

[Brian188]

try this format +1xxxxxxxxxx@incoming.vtnoc.net

Be sure to use the + and your 11 digit phone number.

[kieranmullen]

Thanks but I am trying to forward the calls from viatalk TO my ata which has a different registration ie 401@domain.com

[Brian188]

Oh I see. Can you forward them via the internet forward option in th control panel? It allows forwarding to any sip url.

[kieranmullen]

I did that using the same uri which works for my free ipkall.com number and it doesn't work.

So if it works with ipkall it should work with VT. I submitted a ticket 2 days ago without a response.

km

[Brian188]

Try adding the sip port you are using, if you forwarding to an ATA.

yournumber@sipdomain.com:5061

That is how I forward busy calls to a second line.

[kieranmullen]

thanks but that doesn't make sense when I am using standard ports, they are properly forwarded, ipkallcom works fine & a direct sip call using gizmoproject.com software works fine

km

[kieranmullen]

Sorry I almost forgot. I have it being forwarded to a asterisk server with the firewall off(temporarily) and it has a static IP. So defining the port should not make a difference.

Update: I tried your suggestion. Dialed and still no difference. Perhaps it takes more that 5 minutes for changes in the forwarding to take effect. Any idea how long it might take?

Thanks

KieranMullen

[kieranmullen]

Updated here

http://forums.hostrocket.com/showthread.php?p=110729#post110729

[kieranmullen]

I also tried calling via sip to the number I configured for Internet Forwarding using Free WorldDialup & Gizmoproject.com and they both work.

Still according to viatalk tech its a dns issue and there is nothing then can do even though I can ping the address from multiple locations and 3 other organizations are able to use sip forwarding...

[Brian188]

I also tried calling via sip to the number I configured for Internet Forwarding using Free WorldDialup & Gizmoproject.com and they both work.

Still according to viatalk tech its a dns issue and there is nothing then can do even though I can ping the address from multiple locations and 3 other organizations are able to use sip forwarding...

Have you tried using the IP address instead of the domain name. That will eliminate the DNS issue.

incoming.vtnoc.net resolves to 64.151.85.60 for me.

[kieranmullen]

I dont think we are on the same page here...

Forwarding sets calls to go from viatalk to a sip address out on the internet

So I would not set the sip address to viatalk if I want to get the call out on the internet

However I will try an IP instead of the domain name I am using...

[kieranmullen]

401@71.245.97.60 works with all other providers.. still not working with viatalk.com
perhaps they just want to get rid of the byod users?

[vtjosh]

I have personally looked into this issue. While you may not have believed or liked what Eric had to say during your support call, he was in fact 100% correct.

Let's take a look at the capture during one of your test calls:
================================================
18.503822 64.151.85.68 -> 216.218.201.134 SIP/SDP Request: INVITE sip:+1XXXXXXXXXX@sanfrancisco-1.vtnoc.net:5060, with session description
18.505904 216.218.201.134 -> 64.151.85.68 SIP Status: 100 Trying
Here's one of our incoming servers passing the incoming call along to the sanfrancisco switch you are using.

18.630553 216.218.201.134 -> 71.245.97.60 SIP/SDP Request: INVITE sip:1YYYYYYYYYY@sip2.instantnetworks.net, with session description
Here is our sanfrancisco switch sending you an invite to begin a call

18.669823 71.245.97.60 -> 216.218.201.134 SIP Status: 404 Not Found
Here your server cannot find 1YYYYYYYYYY, or is somehow getting lost in your call logic. Your server then gives up and sends us a 404, basically saying that 1YYYYYYYYYY was not found.

18.669953 216.218.201.134 -> 71.245.97.60 SIP Request: ACK sip:1YYYYYYYYYY@sip2.instantnetworks.net
We acknowledge your 404

18.672670 216.218.201.134 -> 64.151.85.68 SIP Status: 404 Not Found
We send the 404 back to one of our incoming servers

18.736654 64.151.85.68 -> 216.218.201.134 SIP Request: ACK sip:+1XXXXXXXXXX@sanfrancisco-1.vtnoc.net:5060
That server acknowledges the 404 and the call is terminated
================================================
Internet forwarding is working exactly as it should on our end. Otherwise, you would not see the line above forwarding to your domain. This is definitely a configuration issue with your PBX.

Also, to confirm this, I tested internet forwarding with my own Asterisk box and it works perfectly. At the very least, you need something like this in your sip.conf:

[401]
type=peer
username=401
host=dynamic
disallow=all
allow=ulaw
secret=password

Then you can set iforwarding to 401@yourdomain and if you have a phone registered with 401 it will work perfectly. If you are attempting to forward those calls somewhere else, you need an incoming route for 401 that handles any forwarding.

I hope with this information you are able to find the issue on your end and get your PBX working the way you want.

[kieranmullen]

Josh - Can you please simply provide an explanation why 3 other company's have the technology that allow it to work and yet viatalk does not?

KM

[kieranmullen]

FYI I have extension 401 registered with sipbroker.com and e164.org and you can use any of the PSTN numbers from sipbroker.com to make a free call to any ENUM registered number. In this case I registered it through e164.org and surprise it works.

I also was on the IRC channel for freepbx today and someone else wanted to test ENUM in asterisk and so I game them my number XXXXXXX (my home number in ticket) and it worked perfectly.

KM

[vtjosh]

Josh - Can you please simply provide an explanation why 3 other company's have the technology that allow it to work and yet viatalk does not?

KM

This is not a matter of whether or not we "have the technology". Your PBX is not handling the call correctly, that's all there is to it.

Please post your configs for your 401 extension.

[kieranmullen]

Broadvoice
authname=5033849594
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=5033849594
host=sip.broadvoice.com
insecure=very
secret=pohZvpjdFi
type=peer
user=phone
username=5033849594


5033849594@sip.broadvoice.com:pohZvpjdFi:5033849594@sip.broadvoice.com/2063472050

A couple of different points
1)I Allow Anonymous Inbound Sip Calls. That is why anyone can call and anyone does call fine

2)I posted 1503343XXXX@sip2.instantnetworks.net in the trouble ticket. This goes straigh to an ivr in asterisk bypassing any issues there might be with an ATA, firewall etc. The server is a public static ip. We should stick to that instead of 401.

3)I was given sanfrancisco-1a.vtnoc.net however I have now changed it to sanfrancisco-1.vtnoc.net

4)Incomming settings - This is different from 8 other providers. They do not require wildcard matching. Just the number in question.. Why does viatalk do it this way? example

extensions_additional.conf

exten => 15036087920,1,Set(FROM_DID=15036087920)
exten => 15036087920,n,Gosub(app-blacklist-check,s,1)
exten => 15036087920,n,Goto(ivr-2,s,1)

did change to

exten => _15036087920,1,Set(FROM_DID=_15036087920)
exten => _15036087920,n,Gosub(app-blacklist-check,s,1)
exten => _15036087920,n,Goto(ivr-2,s,1)


6) Why does dtmfmode=rfc2833 not work for your system? I was crizied about this setting from the asterisk irc channel :-)

7)All this being said internet forwarding should work without any specific viatalk settings in trunks, outgoing or incomming if you allow annonymos calls. Which I do. So anyone can make a direct call to any extension ivr etc...

???????

8)It (internetforwarding) now works without any change to my IVR or ATA and without adding anything to do with viatalk to the server. Any idea how that would be when the server was getting a 404? There have been no changes to the dns server nor have there been any connection issues...

Something has changed on the vt talk side...

thanks
km

[vtjosh]

In your config on our end, you were iforwarding to 401, so that's why I used it in an example. We use inband DTMF because it works best with the majority of our users.

Anyway, if its working, I'm glad - but nothing changed over here.