Jump to content

pbxnsip basic setup issue - user not found on all calls.


Chad

Recommended Posts

I have a simple voip gateway config on asterisk that I'm trying to get to work on pbxnsip.

 

Environment Info:

192.168.154.135 - pbxnsip w/ 3 minutes demo license

 

Create Domain

Testing1

 

Create Trunk

Name: VoIPgw

Type: SIP Registration

Account: axuserid

Domain: www.axvoice.com

Username: axuserid

Password: axpassword

Outbound Proxy: magnum.axvoice:9060

Accept Redirect: yes

 

Status shows up as 200 OK...Works fine directly on X-lite.

Create Dial Plan

Name: To VoIPgw

Pref Trunk Pattern Replacement

97 VoIPgw 9* *

Create Users

Account Number(s): 500

Dail Plan: To VoIPgw

SIP Password: 500!!@@

PIN: 500

 

Account Number(s): 501

Dail Plan: To VoIPgw

SIP Password: 501!!@@

PIN: 501

 

Configure X-lite softphone

Username: 501

Password: 501!!@@

auth username: 501

domain: 192.168.154.135

proxy: 192.168.154.135

 

Phone connects fine...

 

Test calling user 500

dial 9 500

 

Watching this with WireShark and looking at the log I seem to be getting a 404 not found right after the INVITE gets sent out to pbxnsip. Does anybody see something wrong with this basic configuration?

 

[bEGIN PBXNSIP LOG FILE]

[9] 2009/01/28 20:39:06: SIP Rx udp:192.168.154.1:37792: INVITE sip:9500@192.168.154.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.154.1:37792;branch=z9hG4bK-d8754z-25f04629c0fb9b16-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:501@192.168.154.1:37792>
To: "Example User - 500 (9 500)"<sip:9500@192.168.154.135>
From: "501"<sip:501@192.168.154.135>;tag=8a8fae26
Call-ID: ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1014k stamp 47051
Content-Length: 463

v=0
o=- 9 2 IN IP4 192.168.154.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.154.1
t=0 0
m=audio 44988 RTP/AVP 100 106 0 105 98 8 3 101
a=fmtp:101 0-15
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=alt:1 3 : 2rk1mf+H 1noXwPQV 192.168.1.105 44988
a=alt:2 2 : O/BTupMX gL9apiNx 192.168.154.1 44988
a=alt:3 1 : O0p3nw26 p4zu2Ejh 192.168.5.1 44988
a=sendrecv
[9] 2009/01/28 20:39:06: UDP: Opening socket on port 49592 [9] 2009/01/28 20:39:06: UDP: Opening socket on port 49593 [8] 2009/01/28 20:39:06: Could not find a trunk (1 trunks) [9] 2009/01/28 20:39:06: Resolve 40: aaaa udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 40: a udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 40: udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: SIP Tx udp:192.168.154.1:37792: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.154.1:37792;branch=z9hG4bK-d8754z-25f04629c0fb9b16-1---d8754z-;rport=37792
From: "501" <sip:501@192.168.154.135>;tag=8a8fae26
To: "Example User - 500 (9 500)" <sip:9500@192.168.154.135>;tag=04fbde6860
Call-ID: ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.
CSeq: 1 INVITE
Content-Length: 0

[6] 2009/01/28 20:39:06: Sending RTP for ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.#04fbde6860 to 192.168.154.1:44988 [5] 2009/01/28 20:39:06: Received incoming call without trunk information and user has not been found [9] 2009/01/28 20:39:06: Resolve 41: aaaa udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 41: a udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 41: udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: SIP Tx udp:192.168.154.1:37792: SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.154.1:37792;branch=z9hG4bK-d8754z-25f04629c0fb9b16-1---d8754z-;rport=37792
From: "501" <sip:501@192.168.154.135>;tag=8a8fae26
To: "Example User - 500 (9 500)" <sip:9500@192.168.154.135>;tag=04fbde6860
Call-ID: ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.
CSeq: 1 INVITE
Contact: <sip:9500@192.168.154.135:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.1.2.3120
Content-Length: 0

[9] 2009/01/28 20:39:06: Resolve 42: aaaa udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 42: a udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: Resolve 42: udp 192.168.154.1 37792 [9] 2009/01/28 20:39:06: SIP Tx udp:192.168.154.1:37792: SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.154.1:37792;branch=z9hG4bK-d8754z-25f04629c0fb9b16-1---d8754z-;rport=37792
From: "501" <sip:501@192.168.154.135>;tag=8a8fae26
To: "Example User - 500 (9 500)" <sip:9500@192.168.154.135>;tag=04fbde6860
Call-ID: ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.
CSeq: 1 INVITE
Contact: <sip:9500@192.168.154.135:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.1.2.3120
Content-Length: 0

[9] 2009/01/28 20:39:06: SIP Rx udp:192.168.154.1:37792: ACK sip:9500@192.168.154.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.154.1:37792;branch=z9hG4bK-d8754z-25f04629c0fb9b16-1---d8754z-;rport
To: "Example User - 500 (9 500)" <sip:9500@192.168.154.135>;tag=04fbde6860
From: "501"<sip:501@192.168.154.135>;tag=8a8fae26
Call-ID: ODI2ZDJhNjI0MzRkNmJmZTQ2MWY1MzE0NjQ3ZGZiNjQ.
CSeq: 1 ACK
Content-Length: 0

[9] 2009/01/28 20:39:06: Last message repeated 2 times [9] 2009/01/28 20:39:06: Message repetition, packet dropped

[END PBXNSIP LOG FILE]

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
×
  • Create New...