SOLVED: OpenVZ PIAF external calls don't ring SIP extensions

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vkimball

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I just setup a container using the OpenVZ PIAF image. I've setup an inbound route for calls from Telasip, but if I make the destination of the route either a SIP extension or a ring group, the dialed extensions don't actually ring and (after a pause and ringing heard by the caller) the call goes to the first extension's voicemail.

If I dial the extensions or the ring group manually from another extension, the dialed extension or group does ring.

Any idea what settings I should be looking at?
 
Look at the Asterisk CLI. Is the call ever making it to your Asterisk server? Sounds like a registration problem which could also be a firewall issue.
 
The call is making it to the server. It goes to voice mail on the extension that is the target of the inbound route or to the voice mail of the first extension of the ring group that is the target of the inbound route.
 
The only strange thing I see is the following SIP message in the Asterisk CLI. The From field doesn't seem right:

Reliably Transmitting (NAT) to 172.28.4.109:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.28.4.105:5060;branch=z9hG4bK3e09bb6a;rport
From: "<!DOCTYPE HTML PUBLIC -//IETF//DTD HTML 2.0//EN>
<html><head>
<title>401 Authorization Required</title>
</head><body>
<h1>Authorization Required</h1>
<p>This server could not verify that you
are authorized to access the document
requested. Either you su
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX

Any idea how I can fix that?
 
Okay, I found the problem. Apparently it was a half-configured caller ID superfecta setup.

Everything is working now.
 
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