[asterisk-users] Getting Asterisk 11.5 to use TURN
I've configured TURN in rtp.conf in Asterisk 11.5. The credentials are correct because I can get Chrome to get relay candidates and attach them to the SDP, but Asterisk doesn't want to play ball. There's little documentation -- at least from what I can tell -- on getting TURN working in Asterisk, other than the samples. STUN debug is also of no help, and when I tcpdump the Asterisk box I can see Asterisk trying to reach the TURN server but cannot see any of the packets. If y'all want logs, please just ask for them. Last time I tried to include all of that stuff, your outdated mailing list system choked on it, so it would be best if I only send what's relevant, and I'm not exactly sure what that is, so please bear with me. Thanks! :) James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com - *'How do we convince people that in programming simplicity and clarity—in short: what mathematicians call elegance—are not a dispensable luxury, but a crucial matter that decides between success and failure?' - Edsger W. Dijkstra* -- _ -- 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
[asterisk-users] Asterisk 12 and OPUS Codec
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS codec, which is part of the WebRTC standard as the default codec. Thank you, -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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
[asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1
Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build Asterisk. I'm using the Asterisk pjproject 2.1 port from here: https://github.com/asterisk/pjproject An example of the build errors I'm resolving one by one is this: make[2]: *** No rule to make target `../../pjlib/lib/libpj-x86_64-unknown-linux-gnu.a', needed by `../lib/libpjnath-x86_64-unknown-linux-gnu.a'. Stop. make[1]: *** [/mnt/src/asterisk-11.4.0-rc1/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2 make: *** [res] Error 2 I'm editing the Makefiles and fixing the paths so Asterisk can find the target. For all the people out there smarter than me, is there a better way to go about this? I'm hoping upgrading PJSIP will resolve the crashing issue, and I'll continue going through Makefiles until someone smarter than me can enlighten me. Thank you for your help! -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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
[asterisk-users] Asterisk 11.4.0-rc1 refuses to use the TURN server
After struggling with one way audio issues as a result of STUN binding errors on both the Asterisk side and the Chrome side, we've decided to just simply go with a TURN relay for RTP packets until the issues are resolved. I configured rtp.conf so that all of the STUN related entries are commented out, and I use the following TURN configuration instead: turnaddr=numb.viagenie.ca:3478 ; ; Username used to authenticate with TURN relay server. turnusername=myusername%40gmail.com ; ; Password used to authenticate with TURN relay server. turnpassword=p@ssw0rd I also use the same configuration on the client side. When running a tcpdump, I see that there is traffic to/from the TURN relay: 10.0.1.18.53875 blues.viagenie.ca.nat-stun-port: UDP, length 20 blues.viagenie.ca.nat-stun-port 10.0.1.18.53875: UDP, length 56 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 28 blues.viagenie.ca.nat-stun-port 10.0.1.18.51435: UDP, length 100 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 144 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 144 blues.viagenie.ca.nat-stun-port 10.0.1.18.51435: UDP, length 100 But it's dead silent when doing a tcpdump on the Asterisk server side. The candidates on both sides don't contain relay candidates. Oddly, the client side still has srflx candidates, suggesting STUN is still at work, but the Asterisk side only contains host candidates. Is TURN fully enabled in Asterisk 11? If so, how does one enable it and make it the priority? Thank you, -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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
Re: [asterisk-users] Top Posting
I have an idea! Instead of arguing over whether or not top posting or bottom posting is the way to go, something that obviously no one will *ever*agree on, why not move to Google Groups instead (or something similar to Google Groups). When I post to Doubango's list, it's easy, there's no top or bottom posting wars, it just works. In fact, in a thread, Google Groups usually drops you right to the most recent message, so the people who like top posting can still see the most recent message while the bottom posters will still see the bottom posting format. It's either this, or we can sit and watch intelligent people continue to degrade one another and argue over something with no agreement in site. :) When I mentioned this before, someone from Digium said this will never happen, and it's unfortunate. Maybe they just like to see people bicker and argue. If there's a better alternative to Google Groups, or a way to set preferences in the mailing list so that everyone is happy, maybe that's something that could be done? James On Sun, Dec 30, 2012 at 6:30 PM, Ron Wheeler rwhee...@artifact-software.com wrote: On 30/12/2012 11:13 AM, Patrick Lists wrote: On 12/30/2012 04:26 PM, Ron Wheeler wrote: I participate in a lot of lists and top posting is now the norm since people want to see quickly if the message is worth reading. Isn't it a bit of a stretch to extrapolate your experience with your lists to top posting being the norm? I am subscribed to several lists and bottom posting, proper trimming and commenting inline is the norm there. Actually the norm is determined by the list rules. If the list rules say one must use bottom posting then one should use bottom posting. If someone does not like that then don't subscribe, find another source to ask a question (the forum, LUG, hire a consultant) or just bottom post. Questions come before answers. Answers come after questions. -1 against changing rule #5. Regards, Patrick Not really enough time in the day to keep track of different rules for all the forums. I am more concerned about content than form. As long as the questions get answered, I can figure out where it is but it is a PITA to scroll down through an e-mail to find out that there is nothing there worth reading. I get over 100 e-mails per day that make it through my filters. I like to read the content as soon as it pops up rather than searching for the text. Ron -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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
Re: [asterisk-users] Top Posting
Sorry for double posting, but I realized it was JIRA I spoke with Digium about, not Google Groups and the mailing list... However, I do think it's worth investigating or looking into alternatives that are more user friendly and that can make it easier to communicate with everyone on the list, whether a seasoned pro, top poster, or bottom poster. James On Sun, Dec 30, 2012 at 6:37 PM, James Mortensen james.morten...@voicecurve.com wrote: I have an idea! Instead of arguing over whether or not top posting or bottom posting is the way to go, something that obviously no one will * ever* agree on, why not move to Google Groups instead (or something similar to Google Groups). When I post to Doubango's list, it's easy, there's no top or bottom posting wars, it just works. In fact, in a thread, Google Groups usually drops you right to the most recent message, so the people who like top posting can still see the most recent message while the bottom posters will still see the bottom posting format. It's either this, or we can sit and watch intelligent people continue to degrade one another and argue over something with no agreement in site. :) When I mentioned this before, someone from Digium said this will never happen, and it's unfortunate. Maybe they just like to see people bicker and argue. If there's a better alternative to Google Groups, or a way to set preferences in the mailing list so that everyone is happy, maybe that's something that could be done? 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
Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
://svnview.digium.com/svn/asterisk/branches/11/CHANGES For a full list of changes in the current release, please see the ChangeLog. http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1 Thank you for your continued support of Asterisk! -- _ -- 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 -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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
Re: [asterisk-users] Asterisk 11 WebSockets.
qasimakhan at gmail.com qasimakhan at gmail.com writes: Hi,I was testing with newly introduced websocket support in asterisk 11. I have successfully implemented everything except when i try to make a call i get no audio. I have tried both SipML5 as well as SIP-JS as clients. the call get connected but i never hear any audio stream. I however get the following warning WARNING[2626][C-]: chan_sip.c:9686 process_sdp: Ignoring video stream offer because port number is zero When i turn rtp debug on i can see RTP getting through. CLI Output: http://pastebin.pk/16sip.conf: http://pastebin.pk/17http.conf: http://pastebin.pk/19extensions.conf: http://pastebin.pk/20Regards,Qasim -- _ According to the Asterisk developers, this is an issue in the hands of the browser developers. Here is the wiki page on the Asterisk 11 SIP over WebSockets: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support At this time, no media is flowing. 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
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Hi Sven, I tried out your changes. I had to replace the $_SERVER['REMOTE_ADDR'] with Java's request.getRemoteAddr() since I'm using Jetty not Apache. I got the same results you got, which I also get using the something.invalid header. The peer connects from Chrome, I can dial my cellphone and make it ring, but the Chrome sipml5 client drops the call when the phone starts ringing. When I answer, the cellphone stays connected, but there is no audio. My suggestion is to post your changes to the user interface on the doubango Google Group as it will mean people don't need to modify the code to connect to Asterisk WS. https://groups.google.com/forum/?fromgroups=#!forum/doubango. See if they can incorporate your changes so we don't have to modify the library after each update. As far as the IP address goes, I'm not sure what this is doing since I still see the invalid domain in my SIP traces. James *I did some changes to the sipml5 client and wanted to share this with you guys... Actually only 2 simple changes...*https://github.com/mailsvb/sipml5 *- The main config section has been splitted and made a little more flexible, see *http://i45.tinypic.com/10x59o7.png - Main call.html file has been renamed to .php and some code has been added that will replace the something.invalid with the actual IP of your client PC. Currently I am able to register and at least make my softphone ring ;-) As soon as I answer the outgoing call from sipml5 in the softclient, I get an error in sipml5... You can find my console output here http://pastebin.com/jdkXSMSD I will continue investigating tomorrow... best regards, Sven -- James Mortensen -- _ -- 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
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..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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
Joshua Colp jcolp at digium.com writes: Hi James, I've trimmed the thread down, well, completely. ^_^ From looking at your information and reading the code it looks as though there is a case where this may occur if certain NAT options are enabled. This is certainly a bug as the code should just not execute when WebSocket is involved. For an immediate fix you can set nat=no in the entry in sip.conf. This should change the result and would also explain why this has not been seen by others. Hi Joshua, I'm still getting the same result. Here is what I have in my sip.conf: [general] context=default ; Default context for incoming calls srvlookup=yes port=5060 bindaddr=0.0.0.0 ;pedantic=no rtcachefriends=yes dtmfmode=auto disallow=all allow=g729 allow=ulaw ; Allow codecs in order of preference allow=ilbc ;allow=silk8 allow=gsm ;allow=silk16 ;allow=silk24 ;nat=force_rport nat=no externip=example.org localnet=10.168.151.65/255.255.254.0 qualify=yes [3001] username=3001 secret=x host=dynamic type=friend context=test transport=ws nat=no [3002] username=3002 secret=x host=dynamic type=friend context=test transport=ws nat=no -- _ -- 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
Joshua Colp jcolp at digium.com writes: James Mortensen wrote: Hi Joshua, I'm still getting the same result. I apparently flipped a bit mentally. Try nat=yes for kicks. Otherwise place this info on the issue if not already there and I shall attempt to get to it when I can for a deeper inspection. Hi Joshua, After setting nat=on, force_rport I can now register! However, the calls terminate with a status 488 in the SIP message. I've attached my debug logs here: https://issues.asterisk.org/jira/browse/ASTERISK-20238 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
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=X 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: 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
Andrew Latham lathama at gmail.com writes: On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen james.mortensen at 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 Michael writes: James, can you add this to the issue at https://issues.asterisk.org/jira/browse/ASTERISK-20221 Hi Michael, I apologize. It looks like the patch did fix the crash. I had two instances of Asterisk running, discovered after running ps -C asterisk u. One must have been the previous instance, before applying the patch and recompiling. While the server isn't crashing anymore, I'm still getting the 400 Bad Request and the 401 Unauthorized when trying to login. From what I hear, this could be something that needs to be modified in the sipML5 library itself. Thus, I'll see what I can find. If anyone knows why I'm getting the unauthorized and bad requests, please let me know. I'll update the mailing list as well. Thanks again! 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
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 2012/8/14 James Mortensen james.mortensen at a-cti.com Andrew Latham lathama at gmail.com writes: On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen james.mortensen at 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 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 -- _ -- 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
[asterisk-users] Websockets on Asterisk 11 and SipML5
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=X 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 -- _ -- 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
Re: [asterisk-users] Websockets on Asterisk 11 and SipML5
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=X 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: 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: Sec-WebSocket-Extensions:x-webkit-deflate-frame Sec-WebSocket-Key:UnhnlavzW/Gk6mwJMdLU/w== Sec-WebSocket-Protocol:sip Sec-WebSocket
