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


Chad
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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]

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