On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen <james.morten...@a-cti.com> wrote: > 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). > > > > > > --
James, can you add this to the issue at https://issues.asterisk.org/jira/browse/ASTERISK-20221 -- ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~ -- _____________________________________________________________________ -- 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
