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[SOLVED] Jitsi Softphone Problems

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[SOLVED] Jitsi Softphone Problems

Postby khuff » Mon Sep 24, 2018 12:49 pm

Hey All,

So we just setup a banging new cluster. 17 servers, Master and slave databases, Archive server with terrabytes of space. All going into a local data center. By far the best vici cluster I've ever ran. We got a fresh vicibox install setup on it (8.1), and are in the process of setting it up and getting it ready to go. But we're running into an issue with the softphone and wanted to see if anyone could provide any insight on what could be up.

We're wanting to use load balance logins, but X-lite only let's you have one account and since we're going to end up with 12 different accounts per phone we wanted something with provisioning built in. Came across Jitsi and it marks all our boxes. Was able to setup a provisioning script real quick and had it automaticlly setup all the extensions. However I can't for the life of me get it to nail up. It'll ring but whenever I answer it gives an error:
Code: Select all
call failed "offer contained no valid media descriptions"

I can't find crap on google for that. Thought it is probably some codec issue but I double checked everything and didn't find anything fixed it. Jitsi is setup for uLAW and aLAW (well PCMU/8000 and PCMA/8000 but those are the same I believe), same as the servers. Setup the extension by hand just to be safe and it made no difference. Tested a single phone with xlite and everything works so I assume it is something with Jitsi.

I turned on sip debug and pulled the logs and the only thing I really see is:
Code: Select all
[Sep 24 12:07:34] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:34] SIP/2.0 488 Not Acceptable here


Where 12.xxx.xxx.xxx is my external ip of my workstation. But like I said Jitsi should be using the right codecs.

Anyone familiar with Jitsi and Vicidial have any advice for me? Any idea what I should look into next? Any advice is greatly appreciated. Below are is the full sip debug from nailing up.

Thanks in advance,
Kevin
Code: Select all
[Sep 24 12:07:32] Content-Length: 0
[Sep 24 12:07:32]
[Sep 24 12:07:32] <------------->
[Sep 24 12:07:32] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:32] --- (9 headers 0 lines) ---
[Sep 24 12:07:32] VERBOSE[21729][C-00000008] chan_sip.c: [Sep 24 12:07:32] list_route: hop: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] OPTIONS sip:170.xxx.xxx.xxx SIP/2.0
[Sep 24 12:07:07] VERBOSE[29824] manager.c: [Sep 24 12:07:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 24 12:07:07] VERBOSE[29824] manager.c: [Sep 24 12:07:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 24 12:07:23] VERBOSE[21706] asterisk.c: [Sep 24 12:07:23]     -- Remote UNIX connection
[Sep 24 12:07:32] VERBOSE[29860] manager.c: [Sep 24 12:07:32]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] netsock2.c: [Sep 24 12:07:32]   == Using SIP RTP CoS mark 5
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Audio is at 15168
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Adding codec 100003 (ulaw) to SDP
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Adding codec 100002 (gsm) to SDP
[Sep 24 12:07:23] VERBOSE[21706] asterisk.c: [Sep 24 12:07:23]     -- Remote UNIX connection
[Sep 24 12:07:32] VERBOSE[29860] manager.c: [Sep 24 12:07:32]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] netsock2.c: [Sep 24 12:07:32]   == Using SIP RTP CoS mark 5
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Audio is at 15168
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Adding codec 100003 (ulaw) to SDP
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Adding codec 100002 (gsm) to SDP
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 24 12:07:32] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:32] Reliably Transmitting (NAT) to 12.xxx.xxx.xxx:5060:
[Sep 24 12:07:32] INVITE sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx SIP/2.0
[Sep 24 12:07:32] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport
[Sep 24 12:07:32] Max-Forwards: 70
[Sep 24 12:07:32] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:32] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:32] Contact: <sip:8339999999@170.xxx.xxx.xxx:5060>
[Sep 24 12:07:32] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:32] CSeq: 102 INVITE
[Sep 24 12:07:32] User-Agent: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:32] Date: Mon, 24 Sep 2018 17:07:32 GMT
[Sep 24 12:07:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 24 12:07:32] Supported: replaces, timer
[Sep 24 12:07:32] Remote-Party-ID: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;party=calling;privacy=off;screen=no
[Sep 24 12:07:32] Content-Type: application/sdp
[Sep 24 12:07:32] Content-Length: 266
[Sep 24 12:07:32]
[Sep 24 12:07:32] v=0
[Sep 24 12:07:32] o=root 1779813720 1779813720 IN IP4 170.xxx.xxx.xxx
[Sep 24 12:07:32] s=Asterisk PBX 11.25.3-vici
[Sep 24 12:07:32] c=IN IP4 170.xxx.xxx.xxx
[Sep 24 12:07:32] t=0 0
[Sep 24 12:07:32] m=audio 15168 RTP/AVP 0 3 101
[Sep 24 12:07:32] a=rtpmap:0 PCMU/8000
[Sep 24 12:07:32] a=rtpmap:3 GSM/8000
[Sep 24 12:07:32] a=rtpmap:101 telephone-event/8000
[Sep 24 12:07:32] a=fmtp:101 0-16
[Sep 24 12:07:32] a=ptime:20
[Sep 24 12:07:32] a=sendrecv
[Sep 24 12:07:32]
[Sep 24 12:07:32] ---
[Sep 24 12:07:32] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:32]
[Sep 24 12:07:32] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:32] SIP/2.0 180 Ringing
[Sep 24 12:07:32] CSeq: 102 INVITE
[Sep 24 12:07:32] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:32] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:32] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>;tag=3985b08b
[Sep 24 12:07:32] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport=5060;received=170.xxx.xxx.xxx
[Sep 24 12:07:32] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:32] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:32] Content-Length: 0
[Sep 24 12:07:32]
[Sep 24 12:07:32] <------------->
[Sep 24 12:07:32] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:32] --- (9 headers 0 lines) ---
[Sep 24 12:07:32] VERBOSE[21729][C-00000008] chan_sip.c: [Sep 24 12:07:32] list_route: hop: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:32] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:32]
[Sep 24 12:07:32] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:32] SIP/2.0 180 Ringing
[Sep 24 12:07:32] CSeq: 102 INVITE
[Sep 24 12:07:32] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:32] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:32] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>;tag=3985b08b
[Sep 24 12:07:32] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport=5060;received=170.xxx.xxx.xxx
[Sep 24 12:07:32] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:32] User-Agent: Jitsi2.10.5550Windows 10

[Sep 24 12:07:33] Call-ID: 10b6a79037e8a5b1aaa413cf3c59035c@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=cc58e767
[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-f12d94c539f601b57e51681cca7bad74
[Sep 24 12:07:33] Max-Forwards: 70
[Sep 24 12:07:33] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:33] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:33] Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
[Sep 24 12:07:33] Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------->
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] --- (12 headers 0 lines) ---
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Sending to 12.xxx.xxx.xxx:5060 (NAT)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Looking for s in trunkinbound (domain 170.xxx.xxx.xxx)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- Transmitting (NAT) to 12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] SIP/2.0 404 Not Found
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-f12d94c539f601b57e51681cca7bad74;received=12.xxx.xxx.xxx;rport=5060
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=cc58e767
[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=as4e194964
[Sep 24 12:07:33] Call-ID: 10b6a79037e8a5b1aaa413cf3c59035c@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] Server: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:33] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 24 12:07:33] Supported: replaces, timer
[Sep 24 12:07:33] Accept: application/sdp
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------>
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Scheduling destruction of SIP dialog '10b6a79037e8a5b1aaa413cf3c59035c@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] OPTIONS sip:170.xxx.xxx.xxx SIP/2.0
[Sep 24 12:07:33] Call-ID: bab97f11b509e8e17cedd29e8f71b1da@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=e6d1f884
[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-0ad943f0718233eb32292a4bed375d75
[Sep 24 12:07:33] Max-Forwards: 70
[Sep 24 12:07:33] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:33] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:33] Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
[Sep 24 12:07:33] Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------->
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] --- (12 headers 0 lines) ---
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Sending to 12.xxx.xxx.xxx:5060 (NAT)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Looking for s in trunkinbound (domain 170.xxx.xxx.xxx)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- Transmitting (NAT) to 12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] SIP/2.0 404 Not Found
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-0ad943f0718233eb32292a4bed375d75;received=12.xxx.xxx.xxx;rport=5060
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=e6d1f884

[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=as0971fd2a
[Sep 24 12:07:33] Call-ID: bab97f11b509e8e17cedd29e8f71b1da@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] Server: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:33] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 24 12:07:33] Supported: replaces, timer
[Sep 24 12:07:33] Accept: application/sdp
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------>
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Scheduling destruction of SIP dialog 'bab97f11b509e8e17cedd29e8f71b1da@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] OPTIONS sip:170.xxx.xxx.xxx SIP/2.0
[Sep 24 12:07:33] Call-ID: c7f9b1660d9686c5d6f7a47876d85e5e@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=9eadf33d
[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-76234af748ed63c2c1fd611342e3a56d
[Sep 24 12:07:33] Max-Forwards: 70
[Sep 24 12:07:33] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:33] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:33] Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
[Sep 24 12:07:33] Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------->
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] --- (12 headers 0 lines) ---
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Sending to 12.xxx.xxx.xxx:5060 (NAT)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Looking for s in trunkinbound (domain 170.xxx.xxx.xxx)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- Transmitting (NAT) to 12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] SIP/2.0 404 Not Found
[Sep 24 12:07:33] Via: SIP/2.0/UDP 10.0.1.110:5060;branch=z9hG4bK-373136-76234af748ed63c2c1fd611342e3a56d;received=12.xxx.xxx.xxx;rport=5060
[Sep 24 12:07:33] From: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=9eadf33d
[Sep 24 12:07:33] To: "9998" <sip:9998@170.xxx.xxx.xxx>;tag=as775a0612
[Sep 24 12:07:33] Call-ID: c7f9b1660d9686c5d6f7a47876d85e5e@0:0:0:0:0:0:0:0
[Sep 24 12:07:33] CSeq: 99 OPTIONS
[Sep 24 12:07:33] Server: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:33] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 24 12:07:33] Supported: replaces, timer
[Sep 24 12:07:33] Accept: application/sdp
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------>
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] Scheduling destruction of SIP dialog 'c7f9b1660d9686c5d6f7a47876d85e5e@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33]
[Sep 24 12:07:33] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:33] SIP/2.0 180 Ringing
[Sep 24 12:07:33] CSeq: 102 INVITE
[Sep 24 12:07:33] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:33] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:33] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>;tag=3985b08b
[Sep 24 12:07:33] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport=5060;received=170.xxx.xxx.xxx
[Sep 24 12:07:33] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:33] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:33] Content-Length: 0
[Sep 24 12:07:33]
[Sep 24 12:07:33] <------------->
[Sep 24 12:07:33] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:33] --- (9 headers 0 lines) ---
[Sep 24 12:07:33] VERBOSE[21729][C-00000008] chan_sip.c: [Sep 24 12:07:33] list_route: hop: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:34] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:34]
[Sep 24 12:07:34] <--- SIP read from UDP:12.xxx.xxx.xxx:5060 --->
[Sep 24 12:07:34] SIP/2.0 488 Not Acceptable here
[Sep 24 12:07:34] CSeq: 102 INVITE
[Sep 24 12:07:34] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:34] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:34] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>;tag=3985b08b
[Sep 24 12:07:34] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport=5060;received=170.xxx.xxx.xxx
[Sep 24 12:07:34] Contact: "9998" <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>
[Sep 24 12:07:34] User-Agent: Jitsi2.10.5550Windows 10
[Sep 24 12:07:34] Content-Length: 0
[Sep 24 12:07:34]
[Sep 24 12:07:34] <------------->
[Sep 24 12:07:34] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:34] --- (9 headers 0 lines) ---
[Sep 24 12:07:34] VERBOSE[21729][C-00000008] chan_sip.c: [Sep 24 12:07:34] Transmitting (NAT) to 12.xxx.xxx.xxx:5060:
[Sep 24 12:07:34] ACK sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx SIP/2.0
[Sep 24 12:07:34] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK2377b73e;rport
[Sep 24 12:07:34] Max-Forwards: 70
[Sep 24 12:07:34] From: "ACagcW1537808743654654654654" <sip:8339999999@170.xxx.xxx.xxx>;tag=as333fc4ca
[Sep 24 12:07:34] To: <sip:9998@10.0.1.110:5060;transport=udp;registering_acc=170_xxx_xxx_xxx>;tag=3985b08b
[Sep 24 12:07:34] Contact: <sip:8339999999@170.xxx.xxx.xxx:5060>
[Sep 24 12:07:34] Call-ID: 11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060
[Sep 24 12:07:34] CSeq: 102 ACK
[Sep 24 12:07:34] User-Agent: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:34] Content-Length: 0
[Sep 24 12:07:34]
[Sep 24 12:07:34]
[Sep 24 12:07:34] ---
[Sep 24 12:07:34] VERBOSE[29860][C-00000008] chan_sip.c: [Sep 24 12:07:34] Scheduling destruction of SIP dialog '11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060' in 6400 ms (Method: INVITE)
[Sep 24 12:07:35] VERBOSE[29860] manager.c: [Sep 24 12:07:35]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 24 12:07:38] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:38] Reliably Transmitting (NAT) to 63.76.57.17:5060:
[Sep 24 12:07:38] OPTIONS sip:63.76.57.17 SIP/2.0
[Sep 24 12:07:38] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK1ef2c5fc;rport
[Sep 24 12:07:38] Max-Forwards: 70
[Sep 24 12:07:38] From: "asterisk" <sip:asterisk@170.xxx.xxx.xxx>;tag=as076be98e
[Sep 24 12:07:38] To: <sip:63.76.57.17>
[Sep 24 12:07:38] Contact: <sip:asterisk@170.xxx.xxx.xxx:5060>
[Sep 24 12:07:38] Call-ID: 7c72b11605c9cd900040da7d24c2db5e@170.xxx.xxx.xxx:5060
[Sep 24 12:07:38] CSeq: 102 OPTIONS
[Sep 24 12:07:38] User-Agent: Asterisk PBX 11.25.3-vici
[Sep 24 12:07:38] Date: Mon, 24 Sep 2018 17:07:38 GMT
[Sep 24 12:07:38] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep 24 12:07:38] Supported: replaces, timer
[Sep 24 12:07:38] Content-Length: 0
[Sep 24 12:07:38]
[Sep 24 12:07:38]
[Sep 24 12:07:38] ---
[Sep 24 12:07:38] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:38]
[Sep 24 12:07:38] <--- SIP read from UDP:63.76.57.17:5060 --->
[Sep 24 12:07:38] SIP/2.0 200 OK
[Sep 24 12:07:38] Via: SIP/2.0/UDP 170.xxx.xxx.xxx:5060;branch=z9hG4bK1ef2c5fc;rport=5060
[Sep 24 12:07:38] From: "asterisk" <sip:asterisk@170.xxx.xxx.xxx>;tag=as076be98e
[Sep 24 12:07:38] To: <sip:63.76.57.17>;tag=b5f9f4ebcs
[Sep 24 12:07:38] Call-ID: 7c72b11605c9cd900040da7d24c2db5e@170.xxx.xxx.xxx:5060
[Sep 24 12:07:38] CSeq: 102 OPTIONS
[Sep 24 12:07:38] Server: Brekeke SIP Server rev.408
[Sep 24 12:07:38] Content-Length: 0
[Sep 24 12:07:38]
[Sep 24 12:07:38] <------------->
[Sep 24 12:07:38] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:38] --- (8 headers 0 lines) ---
[Sep 24 12:07:38] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:38] Really destroying SIP dialog '7c72b11605c9cd900040da7d24c2db5e@170.xxx.xxx.xxx:5060' Method: OPTIONS
[Sep 24 12:07:40] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:40] Really destroying SIP dialog '150056d8038617c9d269f696a5324eb4@0:0:0:0:0:0:0:0' Method: OPTIONS
[Sep 24 12:07:40] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:40] Really destroying SIP dialog 'c268f2ac959cfca333a73dff32cd8fa0@0:0:0:0:0:0:0:0' Method: OPTIONS
[Sep 24 12:07:40] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:40] Really destroying SIP dialog '892d0d7269d17f1a37d05693ae61cb61@0:0:0:0:0:0:0:0' Method: OPTIONS
[Sep 24 12:07:41] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:41] Really destroying SIP dialog '11c4c645401ce5f0117e723403208c96@170.xxx.xxx.xxx:5060' Method: INVITE
[Sep 24 12:07:43] VERBOSE[21729] chan_sip.c: [Sep 24 12:07:43] Reliably Transmitting (NAT) to 12.xxx.xxx.xxx:5060:
Last edited by khuff on Tue Sep 25, 2018 10:04 am, edited 1 time in total.
Vicibox 8.1 | ViciDial VERSION: 2.14-687a BUILD: 180908-1618 | Asterisk: 11.25.3-vici | Multi Server 12 x Dialers / 2 x Web / 1 x Master DB / 1 x Slave DB / 1 x Archive | No extra software or hardware after install
khuff
 
Posts: 80
Joined: Mon Feb 20, 2012 12:19 pm
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Re: Jitsi Softphone offer contained no valid media descripti

Postby khuff » Tue Sep 25, 2018 10:03 am

Looks like it is something with my computer. Works on every other machine we've tried it on. +1 for Jitsi btw, the provisioning is great and super easy to setup. In case it helps anyone here's my provisioning script (Codeigniter function), should work as plain ol php if you change the post vars. Just need to add in your ips, and then setup jitsi to point to script location, with username and password for the provisioning.
Code: Select all
https://yourserver.com/jitsi/?user=${username}&password=${password}

It'll then ask for a username and password on start and this will setup the phone extension and registration password for the extensions on each server in your list.
You can also make a batch file that will install the msi package with the provisioning server already set so you just have to run the batch and it'll be ready to go. Just need to make sure the batch file is in the same dir as the msi package. (I used the latest x86 package)
Code: Select all
msiexec.exe /i jitsi-latest-x86.msi  /qb PROV_URL="https://yourserver.com/jitsi/?user=${username}&password=${password}"

Code: Select all
    function jitsi(){
        $user = $this->input->post('user');
        $pass = $this->input->post('password');

        $servers = array(
            "192.168.0.1",
            "192.168.0.2",
            "192.168.0.3",
            "192.168.0.4",
            "192.168.0.5",
            "192.168.0.6",
            "192.168.0.7"
        );
        $counter = 0;

        foreach($servers as $server){
            $counter ++;
            echo "net.java.sip.communicator.impl.gui.accounts.acc0sip$counter=SIP\:$user ($counter)"."\r\n";
            echo "net.java.sip.communicator.impl.gui.accounts.acc0sip$counter.accountIndex=0"."\r\n";
            echo "net.java.sip.communicator.impl.gui.accounts.acc0sip$counter.wizard=net_java_sip_communicator_plugin_sipaccregwizz_SIPAccountRegistrationWizard"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter=acc0sip$counter"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter.ACCOUNT_UID=SIP\:$user"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter.PASSWORD=$pass"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter.PROTOCOL_NAME=SIP"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter.SERVER_ADDRESS=$server"."\r\n";
            echo "net.java.sip.communicator.impl.protocol.sip.acc0sip$counter.USER_ID=$user"."\r\n";
        }
    }
Vicibox 8.1 | ViciDial VERSION: 2.14-687a BUILD: 180908-1618 | Asterisk: 11.25.3-vici | Multi Server 12 x Dialers / 2 x Web / 1 x Master DB / 1 x Slave DB / 1 x Archive | No extra software or hardware after install
khuff
 
Posts: 80
Joined: Mon Feb 20, 2012 12:19 pm
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