James Mortensen <james.mortensen <at> a-cti.com> writes: > > James Mortensen <james.mortensen <at> a-cti.com> writes: > > > > > > > mailsvb <mailsvb <at> gmail.com> writes: > > > > > > > > > > > Hi James, > > > after applying the patch, I got the 400 bad request message as well... > > > This seems to be related to the sipml5 client (same issue with sip-js) > > generating a wrong request. Take a look at the contact header in the REGISTER > > message. > > > > > > I was not able to fix the js code to generate the correct request... In fact > > it should look like this (sip:user <at> local-ip:local-port) > > > > > > regards, > > > Sven > > > > > Hi Sven, > > > > I know this doesn't fix the sipML5 problem, but I changed line 145 of > > tsip_transport.js in the sipML5 library from > > > > return "df7jal23ls0d.invalid"; > > > > to > > > > return "10.x.x.x"; > > > > where 10.x.x.x is the local IP where I'm trying to register from. I am > getting > > a 200 OK from the Asterisk server and am able to connect to it, but I can't > make > > any calls yet. > > > > I'll continue looking at the sipML5 code and will post an update if I get > > anywhere. > > > > Thanks again! > > James > > > > -- > > _____________________________________________________________________ > > Hi Sven, > > According to the developer of sipML5, the problem is that Asterisk 11 doesn't > fully support SIP over WebSockets, which means that the problem is not > necessarily in the sipML5 codebase. > > See the Doubango thread here, as well as the spec Mamadou cites: > https://groups.google.com/forum/?fromgroups#!topic/doubango/jNA0dj5zpKM%5B1- > 25%5D > > He cited the spec, which indicates that the client is supposed to send > "df7jal23ls0d.invalid" as the domain name, since the client side doesn't really > know what to send. > > The main difference I see between the SIP messages in the spec and my SIP > messages is this line: > > I have: > > Supported: path > > Whereas the spec has: > > Supported: path, outbound, gruu > > Anyone know what these do exactly and whether or not sipML5 needs to send > outbound and gruu? > > Thank you, > James > > -- > _____________________________________________________________________
Hello, Here is the error that I'm seeing when trying to register my SIP user from sipML5 to Asterisk 11: <-------------> --- (13 headers 0 lines) --- [Aug 15 17:12:09] ERROR[19510]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Aug 15 17:12:09] WARNING[19510]: chan_sip.c:15314 parse_register_contact: Invalid hostport 'df7jal23ls0d.invalid' [Aug 15 17:12:09] WARNING[19510]: chan_sip.c:16208 register_verify: Failed to parse contact info I also added outbound, gruu to the path header in the SIP message, and this doesn't seem to make a difference. It looks to me like the WebSocket portion of Asterisk is trying to use random ports. Shouldn't it be using the default 8088? The WebSockets spec seems to indicate that WebSockets can use the same port used by the HTTP server, which in this case is 8088. Here is the full output with "sip set debug on": == WebSocket connection from '50.43.101.83:28096' for protocol 'sip' accepted using version '13' <--- SIP read from WS:50.43.101.83:28096 ---> REGISTER sip:50.18.243.242 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKxMm2HP4O04PpG6GFaNUt3v5yGGD6k8Fz;rport From: <sip:3002@50.18.243.242>;tag=TmFBiD1z4TwDR8EvxHSM To: <sip:3002@50.18.243.242> Contact: "3002" <sip:3002@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip- im;+audio;language="en,fr" Call-ID: 1c6c4822-3b14-fb84-c2f8-a1af9b32e2c2 CSeq: 35134 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Organization: Doubango Telecom Supported: path, outbound, gruu <-------------> --- (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=z9hG4bKxMm2HP4O04PpG6GFaNUt3v5yGGD6k8Fz;rport;receiv ed=50.43.101.83 From: <sip:3002@50.18.243.242>;tag=TmFBiD1z4TwDR8EvxHSM To: <sip:3002@50.18.243.242>;tag=as300bd165 Call-ID: 1c6c4822-3b14-fb84-c2f8-a1af9b32e2c2 CSeq: 35134 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="47868ef3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1c6c4822-3b14-fb84-c2f8-a1af9b32e2c2' in 32000 ms (Method: REGISTER) <--- SIP read from WS:50.43.101.83:28096 ---> REGISTER sip:50.18.243.242 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKnpSFmTSk3t1owiThIANGo9cV61xGOfZA;rport From: <sip:3002@50.18.243.242>;tag=TmFBiD1z4TwDR8EvxHSM To: <sip:3002@50.18.243.242> Contact: "3002" <sip:3002@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip- im;+audio;language="en,fr" Call-ID: 1c6c4822-3b14-fb84-c2f8-a1af9b32e2c2 CSeq: 35135 REGISTER Content-Length: 0 Max-Forwards: 70 Authorization: Digest username="3002",realm="asterisk",nonce="47868ef3",uri="sip:50.18.243.242",respon se="d27f92f71e44daed84f7b579546d381b",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Organization: Doubango Telecom Supported: path, outbound, gruu <-------------> --- (13 headers 0 lines) --- [Aug 15 17:43:10] ERROR[19516]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Aug 15 17:43:10] WARNING[19516]: chan_sip.c:15314 parse_register_contact: Invalid hostport 'df7jal23ls0d.invalid' [Aug 15 17:43:10] WARNING[19516]: chan_sip.c:16208 register_verify: Failed to parse contact info <--- Transmitting (no NAT) to 50.43.101.83:5060 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKnpSFmTSk3t1owiThIANGo9cV61xGOfZA;rport;receiv ed=50.43.101.83 From: <sip:3002@50.18.243.242>;tag=TmFBiD1z4TwDR8EvxHSM To: <sip:3002@50.18.243.242>;tag=as300bd165 Call-ID: 1c6c4822-3b14-fb84-c2f8-a1af9b32e2c2 CSeq: 35135 REGISTER Server: Asterisk PBX 11.0.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Date: Wed, 15 Aug 2012 17:43:10 GMT Content-Length: 0 <------------> James -- _____________________________________________________________________ -- 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