[asterisk-users] SIP over WSS connection : mask error
Hi, I use chrome and sipml5 to connect to asterisk webrtc interface using TLS. The wss connection seems ok and the SIP REGISTER sent from chrome to asterisk and the SIP response received. In the response, I get a failed: A server must not mask any frames that it sends to the client error msg and chrome terminates the ws connection. I've checked the asterisk debug logs, and the wireshark tls trace ( after decryption), the mask-bit of the WS Frame being not set. But the chrome complains otherwise. Given below is the chrome error log and the part of the wirehark capture (after decryption). Can someone , help me identify the problem? session event = type = connecting - description = Connecting... test_wss_ast.html:39 session event = type = sent_request - description = REGISTER request successfully sent test_wss_ast.html:39 WebSocket connection to 'wss://nicta-vm1.cloudapp.net:8089/ws' failed: A server must not mask any frames that it sends to the client. SIPml-api.js:3 __tsip_transport_ws_onerror SIPml-api.js:1 __tsip_transport_ws_onclose SIPml-api.js:1 State machine: tsip_dialog_register_Any_2_Terminated_X_transportError SIPml-api.js:1 === REGISTER Dialog terminated === SIPml-api.js:1 session event = type = transport_error - description = Transport error test_wss_ast.html:39 session event = type = terminated - description = Disconnected test_wss_ast.html:39 response code = -1 test_wss_ast.html:41 The FSM is in the final state 009F 81 7e 02 5b 53 49 50 2f 32 2e 30 20 34 30 31 20 .~.[SIP/ 2.0 401 00AF 55 6e 61 75 74 68 6f 72 69 7a 65 64 0d 0a 56 69 Unauthor ized..Vi 00BF 61 3a 20 53 49 50 2f 32 2e 30 2f 57 53 53 20 64 a: SIP/2 .0/WSS d 00CF 66 37 6a 61 6c 32 33 6c 73 30 64 2e 69 6e 76 61 f7jal23l s0d.inva 00DF 6c 69 64 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 lid;bran ch=z9hG4 -- Rgds Thava -- _ -- 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] Bad Magic Internal Error
What does this mean of bad magic internal error, SIP to SIP calling is fine, when I use SIP via GSM I have this, and asterisk restarts automatically. Asterisk version which am using is 11.1.2. Regards -- _ -- 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] VM notification to multiple email recipients
On Wednesday 11 September 2013, Mike Diehl wrote: Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple addresses? Mike Personally, I'd set up a whole new user account to receive the notification e- mails, and then use a procmail recipe to forward it to the necessary accounts. Or if I was feeling particularly lazy, I'd tell the user to go and $ man procmailrc ;) -- AJS Answers come *after* questions. -- _ -- 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] (no subject)
Hi I am running following asterisk installed with apt on Debian 7.1. dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1 amd64Open Source Private Branch Exchange (PBX) ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1 all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm1.4.22-1 all asterisk PBX sound files - en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1 amd64loadable modules for the Asterisk PBX If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won't even send a Trying. We are using* TCP *only. Content-Type: application/sdp Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+. Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- 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] Dealing with muti-body INVITE
Hi I am running following asterisk installed with apt on Debian 7.1. dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1 amd64Open Source Private Branch Exchange (PBX) ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1 all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm1.4.22-1 all asterisk PBX sound files - en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1 amd64loadable modules for the Asterisk PBX If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won't even send a Trying. We are using* TCP *only. Content-Type: application/sdp Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+. Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- 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] Client saw something shiney -- Fonality HUD
An Elastix client saw a Fonality HUD demo and fell in love. I'm not a fan of Fonality as a company, but what do you think of HUD? 1) Does it bring real value? 2) Do I have alternatives? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Index of AGI Scripts
Hi, I wasn't sure quite where to inform people about this, but often I find it hard to find good AGI scripts written by the community, and voip-info is so often out of date. So I create a simple website for people to list their own, or freely available AGI scripts all in one place. http://www.theagigallery.co.uk Any feedback would be welcomed at this time, improvements, issues, general comments etc as I am sure there will be plenty. I've added some that I use and that I've found online already, but I would ask that others add any they know of too, and help build up a nice big free database of AGI scripts for the general Asterisk community. Thanks, Ben -- _ -- 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 get call progress events from WebSocket connected to Asterisk 12 ARI events API
Joshua, Thanks, I am really looking forward to the new REST API support. I know it will take a while to get all the pieces in place. I don't know what the Digium vision is for the REST API, but what I would like to see is a simple WebSocket connection that can receive granular events for all the call activity on the Asterisk server. This would allow a Node.js application to know everything that is happening so it could support UC web apps that also connect to the Node.js server. If the ARI has enough granularity to let the Node.js application make real-time call control decisions and manage call progress and features, then the Asterisk servers(s) could be used as SIP and media edge devices with third party call control running on the Node.js platform. Jim On Thu, Sep 12, 2013 at 10:07 AM, Joshua Colp jc...@digium.com wrote: Jim Fathman wrote: Hello, Bonjour! I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP phones working. Good. I can retrieve data using curl to interact with the new Asterisk REST API (ARI). Good. Now I want to use the new ARI events API, which requires a WebSocket connection. I am using Node.js for the client, and have a stable connection to ARI events on the Asterisk 12 server. What I hope for is that my Node.js client will receive call related events in JSON format messages as call activity occurs on the Asterisk server. But I don't know how to request this information via the API. Do I need to specify something in the query string used for the initial WebSocket connection? Or do I need to send some kind of event subscription messages within the WebSocket once connected? David Lee (ARI man supreme) is currently working on an issue [1] which covers support for subscribing for this information for delivery over the WebSocket connection in a branch [2]. I'd expect this to be integrated into 12 within a few weeks. I believe it should cover what you want to do. [1] https://issues.asterisk.org/**jira/browse/ASTERISK-22451https://issues.asterisk.org/jira/browse/ASTERISK-22451 [2] http://svn.digium.com/svn/**asterisk/team/dlee/ASTERISK-** 22451-ari-subscribe/http://svn.digium.com/svn/asterisk/team/dlee/ASTERISK-22451-ari-subscribe/ Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __**__**_ -- 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 -- _ -- 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 get call progress events from WebSocket connected to Asterisk 12 ARI events API
Jim Fathman wrote: Hello, Bonjour! I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP phones working. Good. I can retrieve data using curl to interact with the new Asterisk REST API (ARI). Good. Now I want to use the new ARI events API, which requires a WebSocket connection. I am using Node.js for the client, and have a stable connection to ARI events on the Asterisk 12 server. What I hope for is that my Node.js client will receive call related events in JSON format messages as call activity occurs on the Asterisk server. But I don't know how to request this information via the API. Do I need to specify something in the query string used for the initial WebSocket connection? Or do I need to send some kind of event subscription messages within the WebSocket once connected? David Lee (ARI man supreme) is currently working on an issue [1] which covers support for subscribing for this information for delivery over the WebSocket connection in a branch [2]. I'd expect this to be integrated into 12 within a few weeks. I believe it should cover what you want to do. [1] https://issues.asterisk.org/jira/browse/ASTERISK-22451 [2] http://svn.digium.com/svn/asterisk/team/dlee/ASTERISK-22451-ari-subscribe/ Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Bad Magic Internal Error
Gopalakrishnan N wrote: What does this mean of bad magic internal error, SIP to SIP calling is fine, when I use SIP via GSM I have this, and asterisk restarts automatically. Asterisk version which am using is 11.1.2. Essentially certain objects within Asterisk have a magic value in them to ensure they are still valid. When you see the above error it means that the object has either had its memory overwritten, or it was freed and is no longer valid. This shouldn't happen. I'd suggest you take a gander at the issue tracker [1] and if no issue matches yours, then create a new one. [1] https://issues.asterisk.org/jira Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] How to get call progress events from WebSocket connected to Asterisk 12 ARI events API
Hello,
I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP
phones working. Good. I can retrieve data using curl to interact with the
new Asterisk REST API (ARI). Good.
Now I want to use the new ARI events API, which requires a WebSocket
connection. I am using Node.js for the client, and have a stable
connection to ARI events on the Asterisk 12 server.
What I hope for is that my Node.js client will receive call related events
in JSON format messages as call activity occurs on the Asterisk server.
But I don't know how to request this information via the API.
Do I need to specify something in the query string used for the initial
WebSocket connection? Or do I need to send some kind of event subscription
messages within the WebSocket once connected?
Any guidance, sample client code, or web reference would be most welcome.
Thanks.
Jim
Node.js client connecting to ARI events:
// app.js
var WebSocket = require('ws');
var ws = new WebSocket('ws://192.168.1.125:8088/ari/events?app=node-client
',
{ headers: {
Authorization: 'Basic Y29tZXQ6MTIzNA=='
},
protocol: 'ari',
});
ws.on('open', function() {
console.log('connected');
});
ws.on('message', function(message) {
console.log('received: %s', message);
});
ws.on('error', function(err) {
console.log(err);
});
It runs and indicates a successful connection:
$ node app.js
connected
The Asterisk CLI logs the successful connection:
== WebSocket connection from '192.168.1.125:34792' for protocol 'ari'
accepted using version '13'
Creating Stasis app 'node-client'
--
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