Chad
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Posts posted by 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]
pbxnsip basic setup issue - user not found on all calls.
in General Setup
Posted
Thanks for the help!