SOLVED New to XIVO - incoming call problems - specific to my voip provider (i think)

mrh

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Hi,

After many years running Elastix on an old server, the hardware has become a bit unreliable, so I thought I'd take the opportunity to move to Xivo on a RPi 3.

The install went fine, and I can make external calls from my extensions - but I can't get incoming calls to behave.

I am with a UK provider called Andrews & Arnold, who's set-up seems slightly different to the examples/tutorials given by Ward (many thanks to Ward and the team for all the Xivo tutorials). I can log on to my voip provider's control panel and see that Xivo has registered fine, but Xivo is not accepting incoming calls!

My Elastix server is set-up as per the guidance at https://support.aa.net.uk/VoIP_Phones_-_FreePBX which states:-

In FreePBX, enter the following:-

USER Context: in-01234567890

USER Details:
type=user
context=from-trunk
username=in-01234567890
remotesecret=YOUR-INCOMING-PASSWORD-HERE
transport=udp
disallow=all
allow=alaw
trustrpid=yes​

It also says:

In order for FreePBX to recognise incoming calls from the voiceless.aa.net.uk platform, we must change a FreePBX setting:

Go to Settings, Asterisk SIP Settings, then click on Chan SIP.

Scroll to the bottom of the page and find the Other SIP Settings entry.

In the first field, enter match_auth_username, and in the second field enter yes.
​
Do I need to do this in Xivo? If so, where do I make this change?

Is anybody else using Andrews and Arnold as their voip provider?

Is there anything unusual about the settings given above?

Many thanks - and apologies for the long post!

Matthew
 
OK, small steps...

I've found "Match users with 'username' field" on the "SIP Protocol properties" and have enabled it. But it still doesn't give me working incoming calls.


My freepbx sip.conf had the following entries......

[in-xxxxxxxxxxx]
type=peer
secret=xxxxxxxxxxxx
insecure=invite
context=from-trunk

[Out-xxxxxxxxxxx]
username=+44xxxxxxxxxx
type=peer
secret=xxxxxxxxxxx
progressinband=no
insecure=invite
host=voiceless.aa.net.uk
context=from-trunk-sip


Should these settings work ok in Xivo, or is it likely that the "+" could be causing a problem?

Thanks

Matthew
 
An update - after a more tinkering I'm still not getting anywhere!

Is anyone able to offer any assistance trying to get XIVO working with my voip provider?

Thanks in advance

Matthew
 
I am not sure it will help you but the setting you were looking for is a checkbox in IPBX > General Settings > SIP Protocol > General > Match users with 'username' field: The 9th item down on the settings.

I have found the most handy thing to do is open a Putty session and use:
Asterisk -vvvvvvr
to watch for incoming calls first to see if they getting thru the firewall and second to see what errors maybe happening.
 
Thanks VaHam,

The Putty idea is a great one. Not sure why I haven't thought of that myself..... I'll give it a try when I get back home this afternoon.

I'll update this thread later.
 
On the Xivo trunks screen, go to the Register tab and try putting the username in the Authorization Username field. Some combination of either no username in the name field and Authorization Username field will probably do it. Play with those two fields and see if you get anywhere.
 
@mrh: Here's some advice from Pascal Cadotte of the XiVO Dev Team:

1. Find a working asterisk sip.conf for the provider (you've already done that)
2. Create a first configuration hoping that it will match the example on step 1. Many examples here.
3. Check the generated configuration for the trunk: xivo-confgen asterisk/sip.conf
4. Change the configuration to account for any missing pieces and repeat step 3...

Once you get it working, please post the formula here for the benefit of others down the road.
 
Thanks for all the suggestions. Just about to sit down at the PC and see what I can find out!

Watch this space!
 
OK - looks like it's something fundamental, as the calls aren't even hitting the cli......

This could end up being something embarrassingly simple!
 
OK. I'll put my hands up. The first issue was firewall related....

Now the calls hit the server and I get:-

NOTICE[2694][C-00000017]: chan_sip.c:26191 handle_request_invite: Failed to authenticate device "xxxxx xxxxxx" <sip:[email protected]>;tag=2016091015443300004

where xxxxx xxxxxx is the mobile number I am making the test call from.

So progress is being made!
 
I'm also getting a dialplan loop (not seen one of these before!):-

[Sep 10 15:59:06] -- Executing [s@from-extern:2] GotoIf("SIP/01234567890-00000010", "1?:not-sip") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:3] GotoIf("SIP/01234567890-00000010", "1?:error-loop") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:4] Set("SIP/01234567890-00000010", "XIVO_DID_NEXT_EXTEN=s") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:5] Set("SIP/01234567890-00000010", "XIVO_FROM_S=1") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:6] Goto("SIP/01234567890-00000010", "from-extern,s,1") in new stack
[Sep 10 15:59:06] -- Goto (from-extern,s,1)
[Sep 10 15:59:06] -- Executing [s@from-extern:1] NoOp("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:2] GotoIf("SIP/01234567890-00000010", "1?:not-sip") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:3] GotoIf("SIP/01234567890-00000010", "0?:error-loop") in new stack
[Sep 10 15:59:06] -- Goto (from-extern,s,10)
[Sep 10 15:59:06] -- Executing [s@from-extern:10] NoOp("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:11] Log("SIP/01234567890-00000010", "ERROR, Dialplan loop detected. Got SIP header To: <sip:[email protected]:5060>") in new stack
[Sep 10 15:59:06] ERROR[8407][C-00000019]: Ext. s:11 @ from-extern: Dialplan loop detected. Got SIP header To: <sip:[email protected]:5060>
[Sep 10 15:59:06] -- Executing [s@from-extern:12] Hangup("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] == Spawn extension (from-extern, s, 12) exited non-zero on 'SIP/01234567890-00000010'
 
OK - seems to have sorted the issues. Will go away and make some notes as to the changes I've made.
 
I'm also getting a dialplan loop (not seen one of these before!):-

[Sep 10 15:59:06] -- Executing [s@from-extern:2] GotoIf("SIP/01234567890-00000010", "1?:not-sip") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:3] GotoIf("SIP/01234567890-00000010", "1?:error-loop") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:4] Set("SIP/01234567890-00000010", "XIVO_DID_NEXT_EXTEN=s") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:5] Set("SIP/01234567890-00000010", "XIVO_FROM_S=1") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:6] Goto("SIP/01234567890-00000010", "from-extern,s,1") in new stack
[Sep 10 15:59:06] -- Goto (from-extern,s,1)
[Sep 10 15:59:06] -- Executing [s@from-extern:1] NoOp("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:2] GotoIf("SIP/01234567890-00000010", "1?:not-sip") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:3] GotoIf("SIP/01234567890-00000010", "0?:error-loop") in new stack
[Sep 10 15:59:06] -- Goto (from-extern,s,10)
[Sep 10 15:59:06] -- Executing [s@from-extern:10] NoOp("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] -- Executing [s@from-extern:11] Log("SIP/01234567890-00000010", "ERROR, Dialplan loop detected. Got SIP header To: <sip:[email protected]:5060>") in new stack
[Sep 10 15:59:06] ERROR[8407][C-00000019]: Ext. s:11 @ from-extern: Dialplan loop detected. Got SIP header To: <sip:[email protected]:5060>
[Sep 10 15:59:06] -- Executing [s@from-extern:12] Hangup("SIP/01234567890-00000010", "") in new stack
[Sep 10 15:59:06] == Spawn extension (from-extern, s, 12) exited non-zero on 'SIP/01234567890-00000010'

What was your solution on this dialplan loop? I am experiencing the exact same while trying to pass a sipuri call through voip.ms registered trunk. Any ideas?
 
Hello,

This is because you don't received the number on the SIP header TO. You only add the number on your contact register input in the trunk sip configuration.
 
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