Asterisk help
Hello all!
I'm a new asterisk user and for a while now I was using a couple of free VoIP providers for inbound and outbound testing. When I started working at another company, one of the perks was that I got a free VOIPo account.
My very first reaction was joy at thinking how cool it'd be to finally deploy my own VoIP gateway for the house. I have a Dell 1U rackmount server that I scored from Goodwill for $30 because it wouldn't post. (they weren't using ECC RAM.)
Well, I got the latest Trixbox/Asterisk/FreePBX ISO image and installed it and have run into a problem. Any VoIP device (softphone, Wifi-Phone, PAP2) can call out from the VOIPo trunk, but any attempt to call in gets a busy signal.
I have checked Google and scoured the Asterisk/FreePBX/Trixbox forums hoping to find a solution but have come up with bupkiss. In fact, some of the settings for the inbound calling, caused me to lose outbound call as well but I was able to get that fixed.
Here's where I need help. I need to find what I'm doing wrong and I hope I'm not the only person with VOIPo that has done this (much less pulled it off). I'm looking forward to getting this thing running and I have great plans for developing it from a testing box that I mess with on occasion to something that I can use full-time.
I have tried entering my credentials into a Zyxel P2000W (WiFi VOIP phone), X-lite for PC, and the PAP2, all work flawlessly both incoming and outgoing, so I know it's not a network issue.
Important information:
Inbound calls fail, but outbound calls work great only when through Asterisk server.
(Perfect quality, even on the old Zyxel Wifi phone.)
{VOIPo assigned #}=Ten digit phone number, no leading "1"
{VOIPo password}= VOIPO Assigned password.
I have "Allow Anonymous Inbound SIP Calls" set to "Yes" but setting it to "No" causes the same symptom and does not change the logfile output.
OS/Asterisk versions:
Asterisk version: Asterisk 1.4.22-3 RPM
CentOS release 5.3 (Final)
Inbound Route settings:
All default, except for the following items
- DID Number = {VOIPo assigned #}
- Destination - Ring Group 600 (all phones), (right now, only ext. 200 exists)
Trunk Settings:
(All fields not mentioned are using their default settings)
- Outbound Caller ID = {VOIPo assigned #}
- Maximum Channels =
- Dial Rules NXXNXXXXXX
- Trunk Name = VOIPo
===Outgoing Settings===
PEER Details:
host=central01.voipwelcome.com
username={VOIPo assigned #}
secret={VOIPo assigned password}
type=peer
context=from-trunk
===Incoming Settings===
USER Context = {VOIPo assigned #}
User Details:
secret={VOIPo password}
type=user
context=from-trunk
disallow=all
allow=ulaw
insecure=very
Registration String:
{VOIPo assigned #}:{VOIPo password}@central01.voipwelcome.com
When I get a call, this is the result of enabling SIP debug (level 10)
Here is a sample of the sip debug info:
<------------>
[Jun 16 12:54:22] VERBOSE[2706] logger.c: Scheduling destruction of SIP dialog 'f4c8cf11-9d306831-59925@67.228.251.106' in 32000 ms (Method: OPTIONS)
[Jun 16 12:54:25] VERBOSE[2706] logger.c: Really destroying SIP dialog '396f1aae60b012da1be594557674bc4f@127.0.0.1' Method: REGISTE R
[Jun 16 12:54:47] VERBOSE[2706] logger.c:
<--- SIP read from 67.228.251.106:5060 --->
INVITE sip:s@192.168.0.14 SIP/2.0
Record-Route: <sip:67.228.251.106;lr=on;ftag=gK0d74ea75;vsf=R1NE dnlmMjhPZklRTmJBTjNHU0R2eWYyOE9mSWo+ETUgHjcyNhcUFQ 8Bf159aX9acXYLfDA0FUQJQFwmCCgjNw-->
Record-Route: <sip:75.126.236.179;lr=on;ftag=gK0d74ea75>
Via: SIP/2.0/UDP 67.228.251.106;branch=z9hG4bKe9e6.3aaa9112.0
Via: SIP/2.0/UDP 75.126.236.179;rport=5060;branch=z9hG4bKe9e6.c271a bc4.0
Via: SIP/2.0/UDP 64.156.174.74:5060;rport=5060;branch=z9hG4bK0dB3f0 a756d33aed219
f: "{Incoming Caller ID Name}" <sip:{Calling Phone #}@64.156.174.74>;tag=gK0d74ea75
t: <sip:{VOIPo Phone #}@75.126.236.179>
i: 386786062_71891924@64.156.174.74
CSeq: 32189 INVITE
Max-Forwards: 68
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIB E,NOTIFY,PRACK,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
m: <sip:{Calling Phone #}@64.156.174.74:5060;nat=yes>
Supported: timer
Session-Expires: 1800
Min-SE: 90
Content-Length: 356
Content-Disposition: session; handling=required
c: application/sdp
v=0
o=Sonus_UAC 15405 4467 IN IP4 64.156.174.74
s=SIP Media Capabilities
c=IN IP4 67.228.251.106
t=0 0
m=audio 62624 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
<------------->
[Jun 16 12:54:47] VERBOSE[2706] logger.c: --- (20 headers 16 lines) ---
[Jun 16 12:54:47] VERBOSE[2706] logger.c: Sending to 67.228.251.106 : 5060 (no NAT)
[Jun 16 12:54:47] VERBOSE[2706] logger.c: Using INVITE request as basis request - 386786062_71891924@64.156.174.74
[Jun 16 12:54:47] VERBOSE[2706] logger.c: Found peer 'VOIPo'
[Jun 16 12:54:47] VERBOSE[2706] logger.c:
<--- 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=z9hG4bKe9e6.3aaa9112.0;recei ved=67.228.251.106
Via: SIP/2.0/UDP 75.126.236.179;rport=5060;branch=z9hG4bKe9e6.c271a bc4.0
Via: SIP/2.0/UDP 64.156.174.74:5060;rport=5060;branch=z9hG4bK0dB3f0 a756d33aed219
From: "{Incoming Caller ID Name}" <sip:{Calling Phone #}@64.156.174.74>;tag=gK0d74ea75
To: <sip:{VOIPo Phone #}@75.126.236.179>;tag=as5840874d
Call-ID: 386786062_71891924@64.156.174.74
CSeq: 32189 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="137a0d35"
Content-Length: 0
<------------>
[Jun 16 12:54:47] VERBOSE[2706] logger.c: Scheduling destruction of SIP dialog '386786062_71891924@64.156.174.74' in 32000 ms (Method: INVITE)
[Jun 16 12:54:47] VERBOSE[2706] logger.c:
<--- SIP read from 67.228.251.106:5060 --->
ACK sip:s@192.168.0.14 SIP/2.0
Via: SIP/2.0/UDP 67.228.251.106;branch=z9hG4bKe9e6.3aaa9112.0
f: "{Incoming Caller ID Name}" <sip:{Calling Phone #}@64.156.174.74>;tag=gK0d74ea75
Call-ID: 386786062_71891924@64.156.174.74
To: <sip:{VOIPo Phone #}@75.126.236.179>;tag=as5840874d
CSeq: 32189 ACK
Max-Forwards: 70
User-Agent: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0
<------------->
[Jun 16 12:54:47] VERBOSE[2706] logger.c: --- (9 headers 0 lines) ---
[Jun 16 12:54:49] VERBOSE[2706] logger.c:
<--- SIP read from 192.168.0.5:49248 --->
I'll try anything, but while I do have the PAP2, I can't afford the digium card needed to bring a POTs line into the server. (theyre about $300!).
Thank you for your help!
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