Routing for inbound trunk not set up properly?

Asterisk Asterisk SIP
1

Hello. I am in process of switching from PRI to a SIP Trunk. I have added the necessary info into sip.conf and extensions.conf. The trunk is registering, however when dialing the number, nothing happens. I attached a debug of the sip peer from asterisk. I am pretty new to asterisk, but looking at the log, what does it look like I am missing? Thank you

<--- SIP read from UDP://55.55.55.55:5060 --->
INVITE sip:8183004595@77.77.77.77 SIP/2.0
Via: SIP/2.0/UDP 55.55.55.55:5060;branch=z9hG4bKg8l4cr30dov1m36nhnj0.1
From: "KNACH,MAR"<sip:818300000@66.66.66.66;user=phone>;tag=1779788950-1532670902896-
To: "acmecorp ."<sip:8183004595@megapathvoice.com>
Call-ID: BW0555028962707181741717748@66.66.66.66
CSeq: 757530937 INVITE
Contact: <sip:8183000000@55.55.55.55:5060;transport=udp>
Diversion: "8183004595 temp"<sip:8182877301@66.66.66.66;user=phone>;reason=unconditional;counter=1
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Call-Info: <sip:66.66.66.66>;appearance-index=1
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Supported:
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 316

v=0
o=BroadWorks 70443717 1 IN IP4 55.55.55.55
s=-
c=IN IP4 55.55.55.55
t=0 0
m=audio 51670 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
a=bsoft: 1 image udptl t38

<------------->
--- (17 headers 15 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 55.55.55.55 : 5060 (no NAT)
Using INVITE request as basis request - BW0555028962707181741717748@66.66.66.66
No user '8183000000' in SIP users list
Found peer 'megapath-sip' for '8183000000' from 55.55.55.55:5060

<--- Reliably Transmitting (NAT) to 55.55.55.55:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 55.55.55.55:5060;branch=z9hG4bKg8l4cr30dov1m36nhnj0.1;received=55.55.55.55
From: "KNACH,MAR"<sip:8183000000@66.66.66.66;user=phone>;tag=1779788950-1532670902896-
To: "acmecorp ."<sip:8183004595@megapathvoice.com>;tag=as52588fea
Call-ID: BW0555028962707181741717748@66.66.66.66
CSeq: 757530937 INVITE
User-Agent: Asterisk PBX 1.6.0.28-samy-r115
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42151dd7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BW0555028962707181741717748@66.66.66.66' in 32000 ms (Method: INVITE)
pbxtra13857*CLI>
<--- SIP read from UDP://55.55.55.55:5060 --->
ACK sip:8183004595@77.77.77.77 SIP/2.0
Via: SIP/2.0/UDP 55.55.55.55:5060;branch=z9hG4bKg8l4cr30dov1m36nhnj0.1
CSeq: 757530937 ACK
From: "KNACH,MAR"<sip:8183000000@66.66.66.66;user=phone>;tag=1779788950-1532670902896-
To: "acmecorp ."<sip:8183004595@megapathvoice.com>;tag=as52588fea
Call-ID: BW0555028962707181741717748@66.66.66.66
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
2

Your peer “megapath-sip” is not configured correctly. You are challenging for authentication and they don’t support that. Your version of Asterisk is also very old and vulnerable to security vulnerabilities, so if you it is publicly exposed you may want to upgrade… or someone could find their way in and place calls.

1 Like
3

Thank you. I thought since the trunk shows registered and I can see the call in debug, that part is set correctly. We have a Cisco ASA in place and allow traffic only via a whitelist, so the world does not have access to the server. I am unable to upgrade because the system backend is tied with Fonality, by upgrading I would probably sever the link to Fonality and break who knows what in the process. Perhaps a project for anotherday.

I just did a quick google and saw that specifying a “secret” turns on authentication. I commented it out, and the call comes through.

You sir are a genius. Thank you VERY much.

4

You should be getting your support from Fonality.

Fonality are in this for money, so they should earn their money. They left a bad taste when they cut off all support, even peer support, for Trixbox.

5

Fonality support is paid, and it is not cheap. We discontinued support about 4-5 years ago. But we own the server and it is on premises, so it continues to work. We have been making changes and tweaks directly to asterisk ourselves. We can’t even get through to an agent without a support contract.