I’m sure everyone’s seen this before, but I’m about at my wits end.
I have an account through VOIPo and while I can call out through X-Lite softphone through the Asterisk server and I can call out via a Zixel P2000w VoIP phone, I can not for the life of me receive any calls, save calling my VoIP number from either the X-lite or the Zyxel.
Incoming phone calls have a long pause at which time ‘sip debug’ goes nuts, but it eventually busies out and nothing happens on the voip phone side.
If I configure either the Zyxel or the X-lite with VOIPo’s SIP settings, I can make and receive calls with no issue whatsoever. The issue comes in when it goes through FreePBX.
My hardware configuration is:
FreePBX CE 2.4.0
Listening IP 192.168.0.25
Kernel Version 2.6.18-53.1.4.el5 (SMP)
Distro Name CentOS release 5 (Final)
It is running inside a VMware installation that for the most part does well. I have the connection bridged, not natted so the 192.168.0 addresses you’re seeing are my actual IP’s.(The idea is I can blast this thing apart, figure out all the little tweaks and whatnot, then I can implement it in an actual server machine later on.)
The Router shouldn’t be a problem as I can see the incoming connection requests and I do have 5000-5003 forwarded to the VMware’s IP.
Incoming Rules:
User Context: (phone number, no dashes or leading digits)
User Details:
context=from-trunk
secret=(password goes here)
type=user
insecure=very
Registration String: (phone number no dashes):(password)@(IP of domain)
[I’ve tried type=user, type=peer, type=friend, canreinvite=no, insecure=invite, nat=yes, nat=no and I’ve tried specifying the host as well with no changes in operation.]
Inbound Routes:
Destination set to ring all phones (600)
[although I have tried sending incoming calls to just the X-lite or the Zyxel with no changes in result]
Per many people’s recommendations, I have looked at sip.conf, but it’s generic and only makes references to other files. If needed I can post replies of those files.
And now for the long part of the message. I did turn on SIP debugging and I made several calls from my cell phone. I was not successful on any attempt.
I apologize for the sheer length, but I want to get this working and I’m hoping someone here can help me with this:
Thanks!
Firestorm_v1
— (7 headers 0 lines) —
Looking for s in from-sip-external (domain 98.200.50.204)
<— Transmitting (no NAT) to 67.23.11.26:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.23.11.26:5060;branch=0;received=67.23.11.26
From: sip:[email protected];tag=c628b177
To: sip:98.200.50.204:5060;tag=as09b95799
Call-ID: [email protected]
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:192.168.0.25
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
<— SIP read from 74.52.58.50:5060 —>
OPTIONS sip:98.200.50.204:5060 SIP/2.0
Via: SIP/2.0/UDP 74.52.58.50:5060;branch=0
From: sip:[email protected];tag=29322287
To: sip:98.200.50.204:5060
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Looking for s in from-sip-external (domain 98.200.50.204)
<— Transmitting (no NAT) to 74.52.58.50:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.52.58.50:5060;branch=0;received=74.52.58.50
From: sip:[email protected];tag=29322287
To: sip:98.200.50.204:5060;tag=as3a11592b
Call-ID: [email protected]
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:192.168.0.25
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
<— SIP read from 75.126.236.178:5060 —>
OPTIONS sip:98.200.50.204:5060 SIP/2.0
Via: SIP/2.0/UDP 75.126.236.178:5060;branch=0
From: sip:[email protected];tag=d60828c3
To: sip:98.200.50.204:5060
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Looking for s in from-sip-external (domain 98.200.50.204)
<— Transmitting (no NAT) to 75.126.236.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 75.126.236.178:5060;branch=0;received=75.126.236.178
From: sip:[email protected];tag=d60828c3
To: sip:98.200.50.204:5060;tag=as2a5027d3
Call-ID: [email protected]
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:192.168.0.25
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
trixbox1*CLI>
<— SIP read from 67.228.251.106:5060 —>
INVITE sip:[email protected] SIP/2.0
Record-Route: sip:67.228.251.106;lr=on;ftag=gK02100e3b;vsf=R1NEdnlmMjhPZklRTmJBTjNHU0R2eWYyOE9mSWo+ETUgHjcyNhcUFQ8Bf159aX9acXYLfDA0FUQJQFwmCCgjNw--
Record-Route: sip:75.126.236.179;lr=on;ftag=gK02100e3b
Via: SIP/2.0/UDP 67.228.251.106;branch=z9hG4bK72c9.24ed6cc7.0
Via: SIP/2.0/UDP 75.126.236.179;rport=5060;branch=z9hG4bK72c9.9619ba25.0
Via: SIP/2.0/UDP 64.158.162.74:5060;rport=5060;branch=z9hG4bK02B52ce209968426386
From: “ILLINGWORTH MAT” sip:[email protected];tag=gK02100e3b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 24772 INVITE
Max-Forwards: 68
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: sip:[email protected]:5060;nat=yes
Supported: timer
Session-Expires: 1800
Min-SE: 90
Content-Length: 356
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 22414 7691 IN IP4 64.158.162.74
s=SIP Media Capabilities
c=IN IP4 67.228.251.106
t=0 0
m=audio 41414 RTP/AVP 0 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:30
a=nortpproxy:yes
<------------->
— (20 headers 16 lines) —
Sending to 67.228.251.106 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘VOIPo’
<— Reliably Transmitting (no NAT) to 67.228.251.106:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 67.228.251.106;branch=z9hG4bK72c9.24ed6cc7.0;received=67.228.251.106
Via: SIP/2.0/UDP 75.126.236.179;rport=5060;branch=z9hG4bK72c9.9619ba25.0
Via: SIP/2.0/UDP 64.158.162.74:5060;rport=5060;branch=z9hG4bK02B52ce209968426386
From: “ILLINGWORTH MAT” sip:[email protected];tag=gK02100e3b
To: sip:[email protected];tag=as2290b6ed
Call-ID: [email protected]
CSeq: 24772 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3cf62fd6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
trixbox1*CLI>
<— SIP read from 67.228.251.106:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 67.228.251.106;branch=z9hG4bK72c9.24ed6cc7.0
From: “ILLINGWORTH MAT” sip:[email protected];tag=gK02100e3b
Call-ID: [email protected]
To: sip:[email protected];tag=as2290b6ed
CSeq: 24772 ACK
Max-Forwards: 70
User-Agent: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0
<------------->
— (9 headers 0 lines) —