SOLVED No incoming or outgoing calls

[jrglass]

I set up PIAF on DigitalOcean and a DID from Vitelity and install X-lite on windows 8.

Both the trunks are registered along with Extension 701.

I cant call in I get a busy and nothing on the CLI

Outgoing can not complete as dialed. I can dial the weather and news from the server
Below are Logs.

Any suggestion whats wrong,

Thanks,

Jeff


Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/701-00000008'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [6142211800@from-internal:1] ResetCDR("SIP/701-00000009", "") in new stack
-- Executing [6142211800@from-internal:2] NoCDR("SIP/701-00000009", "") in new stack
-- Executing [6142211800@from-internal:3] Progress("SIP/701-00000009", "") in new stack
-- Executing [6142211800@from-internal:4] Wait("SIP/701-00000009", "1") in new stack
> 0xb7358c98 -- Probation passed - setting RTP source address to 24.***.**.187:58526
-- Executing [6142211800@from-internal:5] Progress("SIP/701-00000009", "") in new stack
-- Executing [6142211800@from-internal:6] Playback("SIP/701-00000009", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/701-00000009> Playing 'silence/1.gsm' (language 'en')
-- <SIP/701-00000009> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
-- <SIP/701-00000009> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [6142211800@from-internal:7] Wait("SIP/701-00000009", "1") in new stack
-- Executing [6142211800@from-internal:8] Congestion("SIP/701-00000009", "20") in new stack
== Spawn extension (from-internal, 6142211800, 8) exited non-zero on 'SIP/701-00000009'
-- Executing [h@from-internal:1] Hangup("SIP/701-00000009", "") in new stack

== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/701-00000009'



PIAF512*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description
701/701 24.***.**.187 D No No A 1758 OK (38 ms)
vitel-inbound/jrgl_main 64.2.142.26 Auto (No) No 5060 Unmonitored
vitel-outbound/jrgl_main 64.2.142.87 Auto (No) No 5060 Unmonitored
3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]

 

[atsak]

Have you got an outbound route for either . or NXXNXXXXX ???

 

[jrglass]

I have the following 1NXXNXXXXXX; NXXNXXXXXX; NXXXXXX

When making an outgoing call its now asking for a password. The route password is blank.

Thanks,

Jeff

 

[atsak]

All I can tell you is that the call is never getting to the Vitelity trunk, which is usually an issue with outbound route being wrong. Is the Vitelity trunk first in the list in the outbound route?

 

[jrglass]

I have got outbound fixed, thank you for your replies. Now I think I know whats wrong with the incoming. Vitelity has me registering with a different server. I am guessing that I need to add that server to iptable. I dont remember and cant find the commands to edit iptable.

Thanks,

Jeff


PIAF512*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [6144143060@from-sip-external:1] NoOp("SIP/66.241.99.181-00000061", "Received incoming SIP connection from unknown peer to 6144143060") in new stack
-- Executing [6144143060@from-sip-external:2] Set("SIP/66.241.99.181-00000061", "DID=6144143060") in new stack
-- Executing [6144143060@from-sip-external:3] Goto("SIP/66.241.99.181-00000061", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/66.241.99.181-00000061", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/66.241.99.181-00000061", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2014-05-23 16:35:46.996 EDT.
-- Executing [s@from-sip-external:6] Log("SIP/66.241.99.181-00000061", "WARNING,"Rejecting unknown SIP connection from 66.241.99.181"") in new stack
[2014-05-23 16:35:31] WARNING[19892][C-00000044]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 66.241.99.181"
-- Executing [s@from-sip-external:7] Answer("SIP/66.241.99.181-00000061", "") in new stack
> 0xb5291150 -- Probation passed - setting RTP source address to 66.241.99.181:13408
-- Executing [s@from-sip-external:8] Wait("SIP/66.241.99.181-00000061", "2") in new stack
-- Executing [s@from-sip-external:9] Playback("SIP/66.241.99.181-00000061", "ss-noservice") in new stack
-- <SIP/66.241.99.181-00000061> Playing 'ss-noservice.gsm' (language 'en')
== Spawn extension (from-sip-external, s, 9) exited non-zero on 'SIP/66.241.99.181-00000061'
-- Executing [h@from-sip-external:1] Hangup("SIP/66.241.99.181-00000061", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/66.241.99.181-00000061'
PIAF512*CLI>

 

[atsak]

The call is coming through iptables (or you would not see this in the log).
Unknown SIP connection from 66.241.99.181 is the problem here.

The vitality trunk needs to be registered to the server name that is that IP address on your inbound trunk.

You could also allow anonymous SIP connections but a lot of people don't recommend that.

 

[john p]

I think the problem can be resolved by adding to /etc/hosts a line like "66.241.99.181 inbound##.vitelity.net" where ## is the number provided by Vitelity. For some reason, the PBX OS is not properly resolving the IP?name so the PBX does not match the incoming call from an IP to a known source and drops it. You can also allow anonymous SIP but this is a security vulnerability. Hope this helps.

 

[jrglass]

What it turned out to be was I had never set up iptables Once I did it worked.

Thanks,