Voip incoming call does not work

Hello,I need to use a SIP account to receive calls on an extension of my Asterisk. I am using the AsterisNow, and set up through the FreePBX.
If I set up my account at any Voip device I can receive calls without problems, but with account configured on Asterik not receive calls. When making calls with SIP same account I have no problems, only to receive is the problem.
Then describe the configuration as I have in the form of FreePBX, the option Trunks:


PEER
username=user
type=friend
secret=pass
qualify=1000
nat=yes
insecure=port,invite
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
dtmfmode=inband
disallow=all
allow=ulaw&alaw&gsm&g722&g726


USER
user=user
host=sip.voipstunt.com
type=friend
secret=pass
context=from-trunk
insecure=invite


Register String:

user:[email protected]


I clarify that I test with a route incoming calls to take all regardless of the DID to rule out a problem at that stage.
I have set in NAT Settings for SIP, my public IP and the local network.
Look SIP communications that come to my server and nothing appears when I call.
The trunk is recorded in this way:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
enter/user 77.72.169.134 No No 5060 Unmonitored
entrante/user 77.72.169.129 No No 5060 OK (390 ms)

Any suggestions?
Thanks!!!

Well.
You can compare with a sipgate trunk.

username=user
fromuser=user
secret=pass
host=sipgate.co.uk
fromdomain=sipgate.co.uk
type=peer
qualify=yes
nat=yes
insecure=very
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&alaw&g729&gsm&slinear

I have no USER setup just an empty box
and a similar register string to yours.
That works perfect for any Sipgate connection.

Otherwise the same old thing.
Paste some logs of the attempted outbound call so people can see what is happening.

Hv.

Thanks for your help Hv, but still can not make incoming calls. Here is the record of the outgoing call. If this is not what you ask me, if not so tell me how it’s done ?.


192.168.1.36.na-localise > 192.168.1.45.sip: [udp sum ok] SIP, length: 895
INVITE sip:[email protected]:5060 SIP/2.0

192.168.1.36.na-localise > 192.168.1.45.sip: [udp sum ok] SIP, length: 1071
    INVITE sip:[email protected]:5060 SIP/2.0


192.168.1.45.sip > 77.72.169.129.sip: [udp sum ok] SIP, length: 901
INVITE sip:[email protected] SIP/2.0

192.168.1.45.sip > 77.72.169.129.sip: [udp sum ok] SIP, length: 901
    INVITE sip:[email protected] SIP/2.0


192.168.1.45.sip > 77.72.169.129.sip: [udp sum ok] SIP, length: 1097
INVITE sip:[email protected] SIP/2.0

192.168.1.45.sip > 77.72.169.129.sip: [udp sum ok] SIP, length: 1097
    INVITE sip:[email protected] SIP/2.0

192.168.1.45: Asterisk Server
192.168.1.36: phone extension

Thanks again.

I shall drop a line saw Dicko advise

If you look in your freepbx | reports | asterisk logs menu.
Check the log for a call you tried to receive.
in there you will find something like [2016-04-21 00:33:01] VERBOSE[30414] at the start

then use the command
grep “\[30414\]” /var/log/asterisk/full

replacing the 30414 with your own reference.
copy that paste it up and lets see what happens.

I am guessing Dicko will be along shortly anyhow.:slight_smile:

Hv.

Hi Hv, and thanks again. Find a record, but even no record of incoming call, it was with an outgoing call, but not me of the code that you mention it.
The call never enters the asterisk from what I can see.
Any idea what I can do?

Go to asterisk info and see if in sip_peers you can see it registered.

Sounds like either the trunk is not setup/working or there is no catch all inbound route.

or 10,000 other options.

Hv.

into “sip show peers” and I get the following:

Name/username ----------Host---------Dyn---------Forcerport-------Comedia----ACL------Port--------Status-------Description
entrante/user---------77.72.169.129-------------------Yes---------------Yes-------------------5060--------OK (592 ms)

any idea what it means?

thanks!!!

now I see this, it seems that my account is not recorded, but not know why.

Host----------------------------dnsmgr—Username------Refresh------------State-----------Reg.Time
sip.voipstunt.com:5060---------N------nexttics------------120-----------Unregistered

that meens YOU ARR REGISTERED. (EDIT) well the post just above your last.!

So the trunk is on, let me read what your problem was again.!

Gees, that 592 is SLOW.

Anyway, Try to Call your number from anything and see what pops up in the log files.

I think this is the first line of my incoming test call

[2016-04-25 17:30:46] VERBOSE[9543] pbx.c: – Executing [[email protected]:1] Set(“SIP/mytrunkname-0000030d”, “__DIRECTION=INBOUND”) in new stack

See the [9543], use the grep command to search for that (but replace it with your log ref.!).

I have since written a small page to filter the logs.

Hv.

Check how you have setup the trunk.
If they are anything like sipgate leave the incomming section blank and just fill in the Peer stuff above it.
trunk Name
Peer Details and
Register string (below incoming).

Hv.

Hi, I’m trying to configure my account as sip, and still can register.
Set only PEER Details and Register string:

Peer Details:
username=user
type=friend
secret=pass
nat=no
insecure=port,invite
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com

Register String:
user:[email protected]

Attached is a picture with what appears to me:

Thanks for your concern

Try and transplant your details into that config and see what happens
Just change the first 5 (FIVE) lines to your secrets.
Leave the rest as a test
and the register string looks ok BUT some like it to end with /user

Hv.

I just looked at the VoipStunt website. Are you sure you can connect to them with Asterisk? Everything I read seems to indicate that they are an App based connection.

If they are a real VOIP provider, they should have instructions somewhere that tell you how to connect an Asterisk server to the system.

That’s Keen,
I make the mistake of thinking things should work.

Hv.

I’m not saying it won’t, but there should be a guide somewhere on their website that says “connect to asterisk like this!” and I just didn’t see one.

Have not read this.
http://www.voip-info.org/wiki/view/VoipStunt
(!! opens in THIS WINDOW)
Can that be fixed (external links = new tab)

But it had the right words.

Hv.