mailsvb <mailsvb <at> gmail.com> writes: > > > Hi, > > I was facing the very same issue and created a ticket... > > > https://issues.asterisk.org/jira/browse/ASTERISK-20221 > > best regards, > Sven2012/8/13 James Mortensen <james.mortensen <at> a-cti.com> > Andrew Latham <lathama <at> gmail.com> writes: > > > > On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen > > > <james.mortensen <at> a-cti.com> wrote: > > > Hello, > > > > > > I'm trying to register a user using sipml5 on Asterisk 11. I followed the > > > instructions here: > > > http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets > > > > > > I added transport=ws to my sip.conf file: > > > > > > [3002] > > > username=3002 > > > secret=XXXXXXXXX > > > host=dynamic > > > type=friend > > > context=test > > > disallow=all > > > allow=g729 > > > ;allow=all ; Allow codecs in order of preference > > > allow=ilbc > > > allow=silk8 > > > allow=gsm > > > transport=ws > > > > > > > > > I also modified the sipml5 library so that the URL looks like this: > > > ws://example.org:8088/ws with the /ws at the end, as instructed. > > > > > > Now, where I get confused is here: > > > > > > "You will need to change sipml5 to use http://<hostname or IP address of > > > > > > Asterisk>:8088/ws as the URL. WebSocket is only available on the /ws path." > > > > > > > > > Did Joshua mean to say ws:// instead of http://? Because I'm not aware of > > > WebSockets working with http protocols, only ws protocols. Is there > > > something I'm missing here? > > > > > > > > > > > > The error that I'm getting in the sipml5 client is: "Disconnected: Failed > > > to connet to the server" And that typo is not mine. > > > > > > > > > > > > > > > On the server, here is what I see from a tcpdump. The port appears to be > > > open, but I'm not convinced that Asterisk is actually listening for > > > WebSocket traffic: > > > > > > > > > > > > > > > tcpdump -v port 8088 > > > > > > > > > > > > > > > 18:57:03.051712 IP (tos 0x0, ttl 243, id 21320, offset 0, flags [DF], proto > > > TCP (6), length 60) > > > static-50-43-101-83.bvtn.or.frontiernet.net.63036 > > > > ip-10-168-151-65.us-west-1.compute.internal.omniorb: Flags [S], cksum 0x4f7a > > > (correct), seq 4055598050, win 14600, options [mss > > > 1380,sackOK,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop], length > > > 0 > > > 18:57:03.051758 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP > > > (6), length 40) > > > ip-10-168-151-65.us-west-1.compute.internal.omniorb > > > > static-50-43-101-83.bvtn.or.frontiernet.net.63036: Flags [R.], cksum 0xeaf4 > > > (correct), seq 0, ack 4055598051, win 0, length 0 > > > > > > > > > > > > Is there something else I'm missing? Please let me know what additional > > > information you need from me. > > > > > > Thank you! > > > > > > -- > > > James Mortensen > > > > > > > Look to see if the /ws is showing in an "http show status" > > > > ''' > > *CLI> http show status > > HTTP Server Status: > > Prefix: > > Server Enabled and Bound to 0.0.0.0:8088 > > > > Enabled URI's: > > /httpstatus => Asterisk HTTP General Status > > /phoneprov/... => Asterisk HTTP Phone Provisioning Tool > > /amanager => HTML Manager Event Interface w/Digest authentication > > /uploads => HTTP POST mapping > > /arawman => Raw HTTP Manager Event Interface w/Digest authentication > > /manager => HTML Manager Event Interface > > /rawman => Raw HTTP Manager Event Interface > > /static/... => Asterisk HTTP Static Delivery > > /amxml => XML Manager Event Interface w/Digest authentication > > /mxml => XML Manager Event Interface > > /ws => Asterisk HTTP WebSocket > > > > Enabled Redirects: > > / => /static/admin.html > > *CLI> > > ''' > > > Hi Andrew, > I uncommented enabled=yes in http.conf and now see the /ws => Asterisk HTTP > WebSocket. I also modified bindaddr=0.0.0.0 as it was previously 127.0.0.1. I > can connect and I do see the following output in my Chrome NET tab: > Request URL:ws://example.org:8088/ws > Request Method:GET > Status Code:101 Switching Protocols > Request Headersview source > Connection:Upgrade > Host:example.org:8088 > Origin:http://local:8888 > Sec-WebSocket-Extensions:x-webkit-deflate-frame > Sec-WebSocket-Key:fazgtURy132RAFXGRiT9TA== > Sec-WebSocket-Protocol:sip > Sec-WebSocket-Version:13 > Upgrade:websocket > (Key3):00:00:00:00:00:00:00:00 > Response Headersview source > Connection:Upgrade > Sec-WebSocket-Accept:fQA1LFnbYFSxFYAr7Ls1Keh54KY= > Sec-WebSocket-Protocol:sip > Upgrade:websocket > (Challenge Response):00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00 > However, the Asterisk server dies afterwards and must be restarted. The > /var/log/messages file has no helpful information; I was tailing it as I made > one of my connect attempts. > If it helps, I have a local Asterisk 11 setup in verbose mode, and I did see the > following warning message when trying to connect to it instead: > *CLI> [Aug 13 13:17:39] WARNING[567]: res_http_websocket.c:533 > websocket_callback: WebSocket connection from '127.0.0.1:53845' could not be > accepted - no protocols out of 'sip' supported > Also, here is what I see in the Chrome NET tab: (I hope this doesn't confuse > the problem. Keep in mind that these are 2 separate Asterisk 11 instances, one > at example.org and one at 127.0.0.1): > Request URL:ws://127.0.0.1:8088/ws > Request Headersview source > Connection:Upgrade > Cookie:__utma=96992031.124949559.1343691697.1343691697.1343691697.1; > __utmz=96992031.1343691697.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none) > Host:127.0.0.1:8088 > Origin:http://local:8888 > Sec-WebSocket-Extensions:x-webkit-deflate-frame > Sec-WebSocket-Key:UnhnlavzW/Gk6mwJMdLU/w== > Sec-WebSocket-Protocol:sip > Sec-WebSocket-Version:13 > Upgrade:websocket > (Key3):00:00:00:00:00:00:00:00 > Let me know if there is any other information you need. Thanks again for your > help! > James > > -- > _____________________________________________________________________
Hi All, I applied the patch and reinstalled, but the issue with Asterisk terminating still exists. I turned up my debug logging so we could see more of what's happening when I try to connect from the sipml5 client. Before the disconnect, I'm getting a SIP/2.0 401 Unauthorized. The same credentials work when using the sipml5.org:4062 Websocket server. If anyone knows what I'm missing, I'd appreciate some insight. Thanks again! Here is what I see: <--- SIP read from WS:50.43.101.83:2807 ---> REGISTER sip:example.org SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPhRKToHsHAU18BcfrkMkTrEuli21WRNN;rport From: <sip:3...@example.org>;tag=zlmPlTESDfSSHTJ1kJcq To: <sip:3...@example.org> Contact: "Jaymes" <sip:3002@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip- im;+audio;language="en,fr" Call-ID: 15a7e7be-6dda-0d29-3048-c18ae39aa9cb CSeq: 56621 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Organization: Doubango Telecom Supported: path <-------------> --- (12 headers 0 lines) --- <--- Transmitting (no NAT) to 50.43.101.83:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPhRKToHsHAU18BcfrkMkTrEuli21WRNN;rport;receiv ed=50.43.101.83 From: <sip:3...@example.org>;tag=zlmPlTESDfSSHTJ1kJcq To: <sip:3...@example.org>;tag=as76bbb28c Call-ID: 15a7e7be-6dda-0d29-3048-c18ae39aa9cb CSeq: 56621 REGISTER Server: Asterisk PBX 11.0.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ed6b126" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '15a7e7be-6dda-0d29-3048-c18ae39aa9cb' in 32000 ms (Method: REGISTER) <--- SIP read from WS:50.43.101.83:2807 ---> REGISTER sip:example.org SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkI8uGewpCFHwqxCtmUDDwbhTdBNQpNBe;rport From: <sip:3...@example.org>;tag=zlmPlTESDfSSHTJ1kJcq To: <sip:3...@example.org> Contact: "Jaymes" <sip:3002@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip- im;+audio;language="en,fr" Call-ID: 15a7e7be-6dda-0d29-3048-c18ae39aa9cb CSeq: 56622 REGISTER Content-Length: 0 Max-Forwards: 70 Authorization: Digest username="3002",realm="asterisk",nonce="3ed6b126",uri="sip:example.org",response ="96dd23c4248b600374b78307013d9d00",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Organization: Doubango Telecom Supported: path <-------------> --- (13 headers 0 lines) --- ip-10-168-151-65*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). Asterisk ending (0). -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users