Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1
On 03/17/2014 01:56 PM, Eric Wieling wrote: Often it is P-Asserted-ID, but depends on the carrier. You should be asking your carrier how to do this. Be careful, if the carrier doesn't like your CID spoofing they might bill the call to a default number on the account. Speaking as a carrier that allows this, we require the P-Asserted-Identity field. This is the example of a header that we insert with our SBC: P-Asserted-Identity: sip:2325551212@1.2.3.4 The phone number is the identifying marker to tell our Metaswitch the needed information to associate the call to the correct object for billing and call restriction purposes. The IP is the internal IP of our Metaswitch. It is the internal IP due to our MetaSwitch being behind our kamailio SBC. I suspect it is the destination which is rejecting the call because toll free numbers are not considered valid, not your carrier rejecting the call. As a carrier, I have never seen a case where a call (inbound or outbound) was rejected because the received caller ID string contained a toll free number. For me, as long as it passes the number validation step, we are good. And a toll free number looks like any other NAMPA number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic Sent: Monday, March 17, 2014 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1 In a multi-tenant environment, we are sending various CallerIDs outbound from asterisk based on who the user is. We have an insurance agency who would like to present a toll free callerid. This works.. unless they're calling a toll free number. In that case, occasionally, the call fails. However, should we send a correctly formatted npanxx of a local number, the call completes. We have been advised that we can send the billing telephone number as a separate header and the call will complete, all-the-while, presenting the toll free number as the caller id. Does anyone know of the correct header required to provide this functionality? -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 Authentication
Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks -- _ -- 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 Authentication
I am not sure if you understood the problem. Asterisk introduced match_auth_username option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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 Authentication
actually alwaysauthreject was the problem, making it yes was a solution. thanks everyone On Tue, Mar 11, 2014 at 8:17 PM, Eric Wieling ewiel...@nyigc.com wrote: Try setting the sip.conf entry to friend, not peer and not user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Tuesday, March 11, 2014 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Authentication I am not sure if you understood the problem. Asterisk introduced match_auth_username option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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 -- _ -- 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] SPA112 Won't stay up
On 02/06/2014 09:25 AM, Mike Diehl wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. More then likely, the replacement router/modem had a different timeout and it was a luck of the draw that it worked. In routers that allow me to set the UDP timeout, I normally set the timeout to 90 seconds. Most routers that done offer a setting for this are usually set to 90 or 120 seconds Then I usually set my registration time on the ATA's to 60 seconds. The devices I seem to have most issues with are SonicWall routers. Jim Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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
[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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Re: [asterisk-users] VM notification to multiple email recipients
Create an alias on the mail server rather then on each asterisk box. On 09/11/2013 11:44 AM, Mike Diehl wrote: OK, and to make things even more difficult, I store my voicemail and voicemail configuration in MySql. Looks like, for now, I will be creating aliases in /etc/aliases and sync'ing that across my servers Thank you for your suggestions. Mike. On Wed, Sep 11, 2013 at 12:14 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hi You can do this, http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/ If you are using asterisk 1.8 On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.comwrote: 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 Log rotate not working
On 5/21/2013 11:54 AM, Ahmed Munir wrote: Checked in /var/logs/ directory, all logs are not rotating by logrotate. Please advise how can I overcome this issue as I'm using CentoOS 5 Ahmed, Proper log rotation depends on a couple things working together correctly to get the job done. First, you need to make sure you have the space to rotate the logs. If you have compression enabled, logrotate creates a copy of the file(s) as it compresses them. You could be running out of space??? Next you need to verify that everything is in place, follow these steps to do so. Keep in mind that I have CentOS 6.4. So the packages might differ a little in the name and surely in the version numbering. 1) Verify logrotate is installed to your system. # yum install logrotate if it asks you to install it, do so. 2) Verify that crond is installed and running. Below is the output I get when searching yum to see if crond is installed. If your query returns nothing then crond is not installed. [root@jim etc]# yum list all | grep ^cron | grep @ cronie.x86_64 1.4.4-7.el6 @anaconda-CentOS-201303020151.x86_64/6.4 cronie-anacron.x86_64 1.4.4-7.el6 @anaconda-CentOS-201303020151.x86_64/6.4 crontabs.noarch 1.10-33.el6 @anaconda-CentOS-201303020151.x86_64/6.4 If crond is not installed, then you will need to install it. Once you have it installed, move on to the next step. 3) Make sure crond is setup to start at boot time. chkconfig crond on 4) Verify that logrotate is in one of the cron include folders. Mine is located in the cron.daily folder. [root@jim etc]# find /etc/*/logrotate /etc/cron.daily/logrotate If you don't find that the above file exists, you might need to re-install logrotate. Next I would've had you verify that you have a config file in /etc/logrotate.d/ for the asterisk log files. But it seems you already to. After all this, if it still isn't working, double check all the steps above. Let us know if this does or doesn't help. -- Jim Lucas -- _ -- 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 SIP deadlocks - update_provisional_keepalive
On 04/03/2013 08:15 PM, Duane Larson wrote: So it just happened again on both machines at the same time and I was running debug on both servers. I am running OpenSIPS and load balancing between both servers so I am guessing when the invite was sent to the first server it was frozen for some reason and then OpenSIPS sent the invite to the second server and that server was also frozen/deadlocked because of the SIP message. I noticed on both servers the last log that was posted with Asterisk deadlocked was the following Asterisk version 11.0.1 [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11805 instead Asterisk version 11.2.1 [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12423 instead In my last email I posted the debug from the Asterisk server with 11.0.1 version of code. Here is a post of the debug for the Asterisk server with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system A number of deadlocks were fixed in the current release of 11.3. Please read the change log to see if any fit your issue. http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.comwrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 disconnecting SIP Calls after 15 Minutes
On 3/21/2013 12:31 AM, Florian Wolters wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo Florian, As both an VoIP provider and phone system vendor, I had this same problem 2 years ago. In my situation, it turned out that it was nothing to do with either the Asterisk box or the provider. The problem was with a router that we had terminating our T1 connection. As an ISP we provide T1's to many customers and we provide the router as well. In this specific case, the customer purchased a data T1 connection with QoS (sip and rtp) then purchased our IP asterisk phone system with SIP trunks from us as well. The way we found this issue was by switching our the T1 router. Turns out that it fixed the problem. Exact same configuration was on each router. So we started scratching our heads... We then looked at the firmware of the two routers and found that they were different. We provide Cisco 26XX routers. Their are many places on the net talking about the 15 minute NAT timeout issue. If you are not using this device, well, maybe it has a similar bug. -- Jim Lucas -- _ -- 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] Google voice with no voice
On 1/21/2013 7:59 PM, Frank wrote: Actually, the funny thing is that it works randomly. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. In the past, I have had strange behaviors like this as well. Turned out to be a ARP race condition with my firewall with static IP assignments. As soon as the second device would ARP, I would loose connectivity with the first device. Check that you have no other device using the IP address that your D70 is using. Also, make sure that nothing else is competing with the Google Voice registration. -- Jim Lucas -- _ -- 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] DIDForSale spam
On 01/09/2013 10:20 AM, Doug Lytle wrote: I received the same spam myself. No, I did not. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] DIDForSale spam
On 01/09/2013 10:54 AM, jon pounder wrote: On 01/09/2013 01:49 PM, Steve Edwards wrote: I was about to reply 'no' but thought to check my spam logs so now I reply 'yes.' I got a few of them actually. -- _ -- 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 What were the senders IP(s)? -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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
On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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 Dimensioning on newer processors
Where can I find some numbers of Asterisk Dimensioning on newer processor like i7 like number of parallel encoding/decoding sessions. I can see some data on wiki but all are for dated systems. Jim -- _ -- 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] sip-user status
On 12/13/2012 11:39 PM, Hans Witvliet wrote: Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of the other two change to busy. I've seen code/dialplan snippets where you could change your status by dialling a specific extension, on which asterisk will react (and change some variables accordingly), but that is not what i'm looking for. It seems that kamaillo has build-in features to react on sip-simple changes. Can i perform the same trick with asterisk? if so, how? Hans. In * this is done via hints. The devices register with * that they want to be notified when the status of what they want to monitor changes. We, when * knows that it is doing something with the device, * changes the hint status of said device and then sends the notification of status change to the awaiting devices. -- Jim Lucas -- _ -- 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] Hacked by Microsoft?
On 11/28/2012 9:03 PM, jon pounder wrote: On 11/28/2012 11:52 PM, Steve Totaro wrote: You're not serious right ? That is just the center of the country since no better location is available. On Wed, Nov 28, 2012 at 7:45 PM, J Gao j...@veecall.com wrote: This morning someone tried to make sip call through my Asterisk. My server just drop these calls and record them in CDR with IP address: Now I noticed something interesting: The hacker's IP address: 168.63.67.239 whois gave me: NetRange: 168.61.0.0 - 168.63.255.255 CIDR: 168.61.0.0/16, 168.62.0.0/15 OriginAS: NetName:MSFT-EP NetHandle: NET-168-61-0-0-1 Parent: NET-168-0-0-0-0 NetType:Direct Assignment RegDate:2011-06-22 Updated:2012-10-16 Ref:http://whois.arin.net/rest/net/NET-168-61-0-0-1 hmmm Did I just hacked by Micro$oft? Gao http://iplocation.truevue.org/168.63.67.239.html I would put it in the North East. In or around New York. With some questionable routing towards the end of its journey. $ traceroute 168.63.67.239 traceroute to 168.63.67.239 (168.63.67.239), 64 hops max, 40 byte packets 1 49.b167.bendtel.net (66.39.167.49) 0.402 ms 0.345 ms 0.320 ms 2 g0-0-0.c1.sea1.bendtel.net (66.39.191.30) 9.896 ms 9.862 ms 9.919 ms 3 six2.microsoft.com (206.81.80.68) 436.893 ms 297.630 ms 211.67 ms 4 ge-1-3-0-57.wst-64cb-1b.ntwk.msn.net (207.46.46.39) 9.850 ms 9.917 ms 9.909 ms 5 xe-0-2-1-0.co1-96c-1a.ntwk.msn.net (207.46.45.216) 14.10 ms 14.37 ms 13.984 ms 6 ge-7-2-0-0.co1-64c-1b.ntwk.msn.net (207.46.40.166) 14.938 ms 15.28 ms 15.75 ms 7 ge-2-0-0-0.nyc-64cb-1a.ntwk.msn.net (207.46.40.91) 83.664 ms 83.821 ms 83.744 ms 8 207.46.45.231 (207.46.45.231) 172.135 ms 160.999 ms 159.25 ms 9 xe-3-0-0-0.db3-96c-1b.ntwk.msn.net (207.46.42.33) 160.677 ms 158.852 ms 158.812 ms 10 10.22.179.127 (10.22.179.127) 160.594 ms 10.22.178.195 (10.22.178.195) 157.664 ms 10.175.44.3 (10.175.44.3) 160.500 ms 11 10.175.46.247 (10.175.46.247) 159.802 ms 159.636 ms 10.175.46.201 (10.175.46.201) 158.802 ms 12 *^C -- Jim Lucas -- _ -- 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] high capacity analog - sip gateway
On 10/25/2012 01:21 PM, Justin Killen wrote: I'm looking for an fxs- sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- 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 How about this for a setup: 4 port T1 cards (1) Digium TE405P (PCI)~$600 (used) or (1) Digium TE420 (PCI-e 1x)~$1300 (used) and then (4) Adtran Total Access 624 (TA624)~$75 (used) 24 port channel bank We use the TA624's CPE all the time. They are very hard to kill. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] Tell apart between network disruption and asterisk restart via AMI
From AMI you can get uptime. If the uptime is short likely Asterisk restarted. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Oct 19, 2012, at 10:31 AM, Alex Villacís Lasso wrote: I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale. -- _ -- 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 -- _ -- 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] Use of Sangoma D500
Does anyone on the list have any experience with using a Sangoma D500 card with Asterisk to transcode G729? If you could mention pros and cons I would like to hear opinions. Thanks -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] New to Asterisk
Greetings, I am interested in learning more ablout Asterisk. Is there a recommended link for getting started. Can I set up an Asterisk server on my Win 7 local host ?? Is this what I need to do or is there another way of becoming familiar with the Asterisk product ? Any help and guidance for a new user is much appreciated ? Jim -- _ -- 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] Authentication: username and password, also to be from the LAN
Is that now permit and deny are used for. To specify the acceptable IP address(es) the user can connect from? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 26, 2012, at 10:11 AM, bilal ghayyad wrote: Hi All; Is it possible to restrict the authentication to be based on the username and password and to be allowed for IPs within the LAN (for example, 192.168.10.x)? I do not need it to be based on the IP only and do not need it to be based on the username and password only, but I need it to be based on the username password and to be from the specific range, so if the IP address of the client was of the range 192.168.10.x then it is allowede to register with its username and password. No need to specify the IP. If it possible, then is it possible to be a configuration per user? Regards Bilal -- _ -- 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 -- _ -- 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] Finish ChanSpy() when channel spied hangs up
I had submitted a patch some time ago to add option s to chanspy. This would
cause chanspy to exit once the specified change was not longer there. I do not
know if it ever got into a released version as I use ABE. It was not in 1.6 but
might be in 1.8.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 8, 2012, at 4:20 AM, equis software wrote:
I need call to C every time that A call to B, but when A-B hangs up i need to
hang up Asterisk-C call too.
Anyboby know another solution?
On Wed, Mar 7, 2012 at 2:51 PM, equis software equissoftw...@gmail.com
wrote:
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel: SIP/519912@SOFTSWITCH\\nContext:
spy\\nExtension: 23\\nSet:SPYCHANNEL=${CHANNEL} /tmp/${UNIQUEID}.call)
exten = _X.,n,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing/)
exten = _X.,n,Dial(SIP/${EXTEN}@SOFTSWITCH)
[spy]
exten = s,1,Answer
exten = s,2,Chanspy(${SPYCHANNEL}|q)
exten = s,3,Hangup
A call to B
and C (519912) is called by Asterisk to spy the call.
Whe the A-B conversation over, C continue connected to Asterisk, I need
Asterisk hangs up this call.
In my case C is another machine that records the call and can´t hang up when
A-B has finished because it doesn't know.
I don't know if i'm clear
On Wed, Mar 7, 2012 at 1:12 PM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
--
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Mike. Yes sip.ld is the firmware. I wanted to jump in because i saw you had the phantom ringing problem as well. I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did you still have the problem in 3.3.2? I thought I saw in the release notes for 3.3.2 that was resolved. I dont have them infront of me but i suppose it is time to double check as I plan on upgrading 30 phones in the morning. I did test 3.3.2 but the phantom ring seemed so rand i thought i could just no reprouduce it. Thanks!! Jim - Original message - It does update the sip.ld file, yes. So does all upgrades, no? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, February 10, 2012 5:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Did the 4.0.1b update overwrite sip.ld on these phones? If I recall correctly you have to tweak that file to make auto-answer work correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt Sent: Friday, February 10, 2012 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian -- _ -- 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 -- _ -- 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] upgraded 1.8.8.0 10.1.0-rc2: now db warnings
Are you by chance using templates (!) In your sip.con? Ive had access denied errors befor when running as non root. - Original message - I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood of: WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic error or missing database AFAIK, I'm not doing any database puts (or gets). There were no such warnings in 1.8.8.0. What do I need to do to silence these warnings? sean -- _ -- 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 -- _ -- 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] allowguest = yes? no?
What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. That does not mean that strangers can not call an IVR and get to your 1000 extension or even a DID that point right to it. If you are going to allowguest=yes you need to take carfule note of your contexts so as not to allow strangers access to parts of your dial plan that have, lets say long distance routes. Does that help? Thanks!! Jim On 01/24/2012 09:34 AM, Gilles wrote: Hello I don't understand how I should use the allowguest item: If set to yes, callers from the Net should authenticate, but then, how can I allow strangers to call extensions in my system? allowguest If set to no, this disallows guest SIP connections. The default is to allow guest connections. SIP normally requires authentication, but you can accept calls from users who do not support authentication (i.e., do not have a secret field defined).Certain SIP appliances (such as the Cisco Call Manager v4.1) do not support authentication, so they will not be able to connect if you set allowguest=no: allowguest=no|yes (from Asterisk – The future of Telephony) Thank you. -- _ -- 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 -- _ -- 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] Force CDR to be written.
Is there a way to Force the CDR data to be written prior to Hanging up the channel? Thanks!! Jim -- _ -- 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] Call abandoned from queue not showing in CDR (possible bug)
I think in your cdr.conf you are looking for the unanswered= directive. Thanks!! Jim - Original message - Hi I'm using 1.8.7.0 with the RealTime architecture. If a call goes into application Queue and is abandoned by the caller, no entry is made in the CDR. Entries are made into the queue log. This cannot be correct behaviour, all calls should show in the CDR. Could anyone else try to reproduce this and if others get the same thing, I'll raise a bug on it. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 -- _ -- 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] Problems with codec translation when using Monitor and MixMonitor
Here is a matrix we put together about g729 license needs: == = == === Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders == = == === ulaw ulaw yes yesyes00 ulaw ulaw yes yesno 00 ulaw ulaw yes no no 00 ulaw ulaw yes no yes00 ulaw ulaw no yesyes00 ulaw ulaw no yesno 00 ulaw ulaw no no no 00 ulaw ulaw no no yes00 ulaw g729 yes yesyes33 ulaw g729 yes yesno 23 ulaw g729 yes no no 11 ulaw g729 yes no yes33 ulaw g729 no yesyes33 ulaw g729 no yesno 23 ulaw g729 no no no 11 ulaw g729 no no yes33 g729 ulaw yes yesyes25 g729 ulaw yes yesno 25 g729 ulaw yes no no 11 g729 ulaw yes no yes23 g729 ulaw no yesyes25 g729 ulaw no yesno 25 g729 ulaw no no no 11 g729 ulaw no no yes23 g729 g729 yes yesyes47 g729 g729 yes yesno 37 g729 g729 yes no no 11 g729 g729 yes no yes45 g729 g729 no yesyes47 g729 g729 no yesno 37 g729 g729 no no no 11 g729 g729 no no yes45 -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote: On 01/12/2012 11:57 AM, Daniel - Asterisk wrote: The simplest answer, I purchased one additional license and one simultaneous call is being recorded now. I do not understand why the ulaw codec (or format) is involved here (... No translator path from alaw to unknown ...) Any entry will be very
Re: [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1
One good thing is now that you know what the problem is you should be able to work with zopier support and get them to fix zopier. They have been very responsive to a couple problems I have and I am running the free version. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2012, at 3:03 PM, Kevin P. Fleming wrote: On 01/12/2012 11:58 AM, Alex Villacís Lasso wrote: I have discovered the root cause of the issue. Due to a peculiarity of Zoiper 2.18, this program will *not* send a ACCEPT or RINGING packet back to Asterisk unless the NEW packet that announces the incoming call contains an IAX_IE_CALLING_NUMBER information element. It does not matter if the calling number is empty, but the corresponding IE must exist. This behavior is a change between Asterisk 1.6 and Asterisk 1.8. Well, I applaud your troubleshooting skills and analysis... well done! Unfortunately, that IE is *not* mandatory in an IAX2 NEW packet, and thus Zoiper failing to properly process such NEW packets is a bug in Zoiper. Yes, Asterisk's behavior has changed (since Caller ID handling was overhauled in Asterisk 1.8, while adding Connected ID support), but both the old and new behavior are compliant with the IAX2 protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- _ -- 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] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
I think the wiki may have just missed func_curl. I have a couple 1.8.x machines with working func_curl. Have you tried to compile it anyway? Thanks!! - Original message - Hi, I've seen that function CURL is missing from 1.8 but back in with 10 (see wiki.asterisk.org). With asterisk 1.8 and above, for a custom CID Name lookup application, which is the most efficient way to send an HTTP GET from the dialplan and parse its response (code and content) ? 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 -- _ -- 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] Best non polycom SIP conference room phone
You got me. At first the polycom world was hard to get into. But with a little effort to understand the configs and the joys of central provisioning the Polycom are my go to endpoint. Couple the endless configurablity with Polycom quilaty and I have many happy clients. As an aside is that what they use on the dCap? I have been meaning to get that when time allows. Thanks!! Jim - Original message - Luke Hamburg l...@solvent-llc.com writes: Carlos- Sorry if this is too much of a digression but this piqued my interest as I've been pretty happy with Polycom in my limited experience (haven't used the SPAs much, just Yealink Polycom, and an occasional Snom here and there). If the config files were not the issue for you, then what _were_ the problems? A button has been pressed. Polycom must reboot for the change to take effect. Reboot now (Y/N)?. Yes it's a recycled Windows joke, but it applies much better to Polycom than it did to Windows. It is IMHO a bit mean to use Polycom's in the Asterisk exam; the difficulty of passing the exam is quite high if you haven't worked with them before. Pretty much anything else is quicker to get to basic working state. Of course, once you get provisioning working they are excellent phones. /Benny -- _ -- 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 -- _ -- 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] Answering call from queue, then put back in queue?
One way to deal with this is to have two queues. Give priority to the original queue callers land in. Once answered put the call in to the second queue. They will then be in the second queue in the order the agents answered the first queue. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote: Version: Asterisk 1.8.x Question: Is it possible for an agent to answer a call from a queue, then place the call back in the queue in the same position they were in? Seems that the answer would be yes to the remove from queue, then place back in by having the agent just transfer the call back to the queue but is there any way to put them back in line where they were? The idea is that the owner of the queue doesn't want callers waiting on hold without first having an agent at least answer the call and ask them to please hold. What's the best way to handle this? Thanks in advance! --Todd -- _ -- 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 -- _ -- 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] calling specific 1800-number not going through.
It took 36 seconds for that number to answer when I called it and it looks like
the call hung up after 32000 ms when you dialed via asterisk.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 5, 2012, at 5:45 PM, Joseph wrote:
I have a strange problem.
I'm using the same dialplan to call 1800-number:
[toll-free]
;second 7 audiocodes strips
exten = _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)
When I call this number (through pstn-5665) 18005000347 the phone always
rings busy.
When I call any other 1800-number the calls goes through.
When I call the same phone number 18005000347 through a different line the
calls goes through every time.
Here is call (busy) trace to that 18005000347 with sip debug ON:
Can anybody decipher why I'm getting busy signal to that particular
1800-number but not others?
--- SIP read from UDP:10.0.0.110:5060 ---
OPTIONS sip:gateway@10.0.0.110 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
Max-Forwards: 70
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Contact: sip:gateway@10.0.0.110:5060
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0
-
--- (12 headers 0 lines) ---
Looking for gateway in default (domain 10.0.0.110)
--- Transmitting (NAT) to 10.0.0.110:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110;tag=as7091ae01
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in
32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 81.15.150.20:5060:
OPTIONS sip:sip.actio.pl SIP/2.0
Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
Max-Forwards: 70
From: asterisk sip:asterisk@10.0.0.100;tag=as64f6417c
To: sip:sip.actio.pl
Contact: sip:asterisk@10.0.0.100:5060
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
User-Agent: Centrala
Date: Fri, 06 Jan 2012 01:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
--- SIP read from UDP:81.15.150.20:5060 ---
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP
10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
To: sip:sip.actio.pl;tag=4fc8ac12
From: asterisksip:asterisk@10.0.0.100;tag=as64f6417c
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
Content-Length: 0
-
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'66070317301f64861df62d20769ba385@10.0.0.100:5060' Method: OPTIONS
-- Accepted AUTHENTICATED TBD call from 10.0.0.108
--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0
-
--- (12 headers 0 lines) ---
Sending to 10.0.0.110:5060 (NAT)
--- Transmitting (no NAT) to 10.0.0.110:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3a451a5b
Content-Length: 0
Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000
ms (Method: REGISTER)
--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Authorization: Digest
username=11,realm=asterisk,nonce=3a451a5b,uri=sip:10.0.0.100,algorithm=MD5,response=5dd6df18064f3d23cb86ca306820e596
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow:
REGISTER,OPTIONS
Re: [asterisk-users] Question on system command 1.4.43
Does the user Asterisk is running as have access to this device?
Thanks!!
- Original message -
I have a USB to serial converter attached to my box. pl2303: Prolific
PL2303 USB to serial adaptor driver
if I login to the box and send/receive serial commands over this unit it
works without error EVERY time.
however, if I run the same command set from with-in the extensions.conf
with System()
I get errors in dmesg like pl2303 ttyUSB0: pl2303_open - failed
submitting read urb, error -22
and then obviusly my command does not work.
however, again - If I go back to the command line and my command it runs
just fine.
Any ideas on what might be happening here?
Jerry
extensions.conf
exten = s,1,ChanIsAvail(Console/Dsp)
exten = s,n,GotoIf($[${AVAILCHAN} = ]?smvoice-busy,s,1)
exten = s,n,System(/home/silentm/bin/usbserial -start)
exten = s,n,Playback(beep)
exten = s,n,Dial(Console/dsp)
exten = s,n,Hangup
exten = h,1,System(/home/silentm/bin/usbserial -stop)
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Re: [asterisk-users] Problem w/ PC port on Polycom 335
Agreed. Check the switch for some kind of port security. Most of the time this would disable the interface if more than one MAC is present but you never know. Are there blinky lights on the pc? Also if provisioning via some sort of server check the MAC-boot log that the pgone uploads. Good Luck!! Thanks!! Jim. - Original message - Mike Diehl wrote: Usually, it just works... Any ideas? I've seen this before. One of our facilities have 'smart or managed' switches that have caused no ends of problems, including preventing computers plugged into the phones not having network access. You may want to review your switches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. Thanks -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Yes, you are missing the fact that the verbose setting controls what level of output will be generated in the first place. You can raise and lower the amount of stuff logged/printed on CLI. The lines in logger.conf control what types of lines go to which place. One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:24 PM, Bruce B wrote: Okay, but I thought that the line console = is supposed to be for CLI and the line Full = is supposed to be for the file /var/log/asterisk/full. Why would the Full = be effected by core set verbose 0? Is this just bad assumption on the part of the developers? I would only assume that core set verbose 0 should only effect what I see at CLI level and not at my my /var/log/asterisk/full log file. Am I missing something? Thanks for the feedback. On Fri, Dec 30, 2011 at 6:19 PM, Jim Dickenson dicken...@cfmc.com wrote: If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. Thanks -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 4:55 PM, Bruce B wrote: One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? No the verbose command controls how much verbose stuff is output. The debug command controls how much debug stuff is output. These are absolute controls of that information. As I said in my original email you can turn off stuff going to the CLI with the logger mute command. That way you do not adjust the verbose level at all. You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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 -- _ -- 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] func_odbc not returning whole smalldatetime MS Sql field.
Hey All,
Odd thing. I am just trying to return the whole date time stamp from a
SMALLDATETIME field in a MS SQL server.
func_odbc.conf = readsql=SELECT DateCreated FROM [REDACTED] WHERE Code
= '${ARG1}'
Problem is I only get the first 15 back from the field. Like so...
Connected to Asterisk 1.8.6.0 currently running on [REDACTED]-dev (pid =
2240)
Verbosity is at least 3
[REDACTED]-dev*CLI odbc read ODBC_[REDACTED]-LOOKUP 104809 exec
DateCreated 2011-12-19 13:2
Returned 1 row. Query executed on handle 0 [asterisk-mssql-connector]
Notice how it only returns 2011-12-19 13:2 and not the rest of the
time... I have run the query on the SQL server and then from isql and it
works everytime leaving the only abstraction point Asterisk.
Any thoughts?
Thanks!!
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Re: [asterisk-users] how to listen on different sip port for a device?
Why not use IAX trunk instead of SIP. This would make it very easy to talk between the two * systems. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 26, 2011, at 4:07 PM, sean darcy wrote: On 12/26/2011 05:43 PM, Yaroslav Panych wrote: 2011/12/26 sean darcyseandar...@gmail.com: So how do I get * to listen to two different ports? sip.conf section [general] bindport=whatever-port-you-want Thanks, but the problem is to get more than 1 port, 5060 and (at least) one other. sean -- _ -- 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 -- _ -- 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] GOIP GSM to SIP Gateway?
I would think it would be better to set a variable for each user and then have
a single context with something like:
_NXX,1,Dial(SIP/${WhatToUse}/${EXTEN})
Or something like this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 20, 2011, at 1:03 PM, John Kiniston wrote:
On Tue, Dec 20, 2011 at 12:39 PM, Matt mhop...@gmail.com wrote:
Is there anyway (short of defining dial an 8 from this phone for this
trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
get it to use certain SIM cards when calls are made from certain
phones?
You could define multiple contexts with different pattern matches for each
GSM connection and and set your phones to use them, phones 1-3 in context1,
phones 4-6 in context2, etc.
[context1]
_NXX,1,Dial(SIP/GSM1/${EXTEN})
[context2]
_NXX,1,Dial(SIP/GSM2/${EXTEN})
[context3]
_NXX,1,Dial(SIP/GSM3/${EXTEN})
--
A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build
a wall, set a bone, comfort the dying, take orders, give orders, cooperate,
act alone, solve equations, analyze a new problem, pitch manure, program a
computer, cook a tasty meal, fight efficiently, die gallantly. Specialization
is for insects.
---Heinlein
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Re: [asterisk-users] Play audio file for both Caller and Callee in a call
You also use AMI to inject audio into the conversation using the ChanSpy application. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote: You can’t per se, but you can call an AGI using stream? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of c.savinov...@itntelecom.com Sent: Thursday, December 15, 2011 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation? Thanks C. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, December 15, 2011 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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 -- _ -- 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 -- _ -- 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] Play audio file for both Caller and Callee in a call
Use an AMI packet like this:
Action: Originate
Channel: Local/do_playback@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280
ActionID: PlayBack
Async: true
With dialplan like this:
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
You need to issue an AMI packet for each leg of the call. Each leg will hear
the same audio feed offset by however long it takes the packets to be
processed. In general this is a few milliseconds and should not be a big deal.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 10:27 PM, virendra bhati wrote:
Hi,
Plese give a little example of script so that it will be clear.
On Thu, Dec 15, 2011 at 11:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
You also use AMI to inject audio into the conversation using the ChanSpy
application.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote:
You can’t per se, but you can call an AGI using stream?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
c.savinov...@itntelecom.com
Sent: Thursday, December 15, 2011 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in
a call
Dear Danny:
How can you use Playback in the middle of 2 channels engaged in a
conversation?
Thanks
C. Savinovich
Original Message
Subject: Re: [asterisk-users] Play audio file for both Caller and
Callee in a call
From: Danny Nicholas da...@debsinc.com
Date: Thu, December 15, 2011 9:31 am
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Playback? What flavor of Asterisk are you using?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS
ARNAL
Sent: Thursday, December 15, 2011 10:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play audio file for both Caller and Callee in a
call
Dear all,
Anyone of you knows how to play an audio file at the beginning of a call for
both Caller and Callee?
A(x) of Dial application only plays audio for callee. I don’t want to use
MeetMe because I want to use Monitor and MixMonitor.
Thank you!
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
nuestra política de envío y recepción de correo electrónico en el enlace
situado más abajo.
This message is intended exclusively for its addressee. We only send and
receive email on the basis of the terms set out at.
http://www.tid.es/ES/PAGINAS/disclaimer.aspx
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
Re: [asterisk-users] A new hack?
On 11/26/2011 5:00 PM, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful to others. Why is that? Even if they don't compromise an account they are still using your bandwidth and resources on your machine. How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? Also, since both methods involve the use of iptables, where exactly is the bandwidth savings? -- Jim Lucas -- _ -- 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] AMI: anything to glue originate to events?
The easiest thing to do is to create userevents in your dialplan to passed to AMI details you want to key off of. In the original originate you can set so variable that you pass to various macros and what have you. These then generate userevents that AMI can use to track the flow of the call. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 17, 2011, at 12:02 PM, giovanni.v wrote: On 17/11/2011 19.45, c.savinov...@itntelecom.com wrote: if it is what I think it is, I remember I had a similar situation a few years ago, and I ended up having to create an internal table in my code, so that I could keep track of the channel ids + action ids . Which is exactly what I'm doing but I tried to figure out if there was something more reliable ... I refer to the logic not the data structure. Ignoring for a moment the relationship between events, my conclusion is still that there is nothing that ensures that very first event that I will receive after /Response/ to my /Originate/ for that channel is really fired from my application, I can only guess it is. Thank you. -- _ -- 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 -- _ -- 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] 2 pbxes
Yes. If you have two asterisk boxes running you can trunk them together and place calls from one to to the other. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 3, 2011, at 11:36 AM, mattias wrote: if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -- _ -- 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 -- _ -- 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] Binding asterisk to two static IPs
On 10/12/2011 3:55 PM, ge...@riseup.net wrote: After reading your original message, this is clear, yes. Sorry for being sloppy. np ;) Anyone else? Would be really really great... I solved it by having two physical connections to my network. PBX E0 IP 192.168.100.36 NM 255.255.255.0 GW 192.168.100.1 E1 IP 192.168.101.254 NM 255.255.255.0 GW n/a All the phones reside withing the 192.168.101.0/24 network. I still have bindaddr=0.0.0.0 so I can talk to my provider and my phones. But on two different interfaces. That forces the communication to always come from the correct source IP addr. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ http://www.bendsource.com/ C - (541) 408-5189 O - (541) 323-9113 H - (541) 323-4219 -- _ -- 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] permit -- deny not working
I do not know if order is important but I always deny all then permit what I want to permit. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Oct 11, 2011, at 1:15 PM, hussein korbani wrote: Hello, i am having an issue with the DENY permit thingy in the Extensions.conf whenever i use the permit deny , all the calls coming from another sip-trunk to my asterisk ,start to fail doesn't use the Extensions dial plans that i created my context contain the following: [context1] . host=1.2.3.4 permit=1.2.3.4/255.255.255.255 deny=0.0.0.0/0.0.0.0 .. can anyone explain why its failing? P.S; calls from extension created on the ASterisk works fine even when permit\deny is used or not Best regards, Hussein K -- _ -- 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 -- _ -- 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] Add SIP diversion header in originate from AMI?
You can dial a local channel which executes a dial plan that does what you want.
Channel: Local/dial_number@cfmc_cdi_private
This will use exten dial_number in the cfmc_cdi_private context.
If you add something like this to the originate packet
Variable: CfMC_Use_CID=5419712513
You can use ${CfMC_Use_CID} to get the value.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 7, 2011, at 8:03 AM, Tobias Steen wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected to
an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to dothis
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls
I do not know when the recording actually starts but if it start when the agent
answers the call then it might be possible to have the name set in an AGI that
gets run when the agent answers call. If nothing else you can set a variable to
the name you want to have the file have and rename it at end of call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2011, at 10:30 PM, Sam Govind wrote:
:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue variables are not populated by then so filename won't contain the Agent
Number.
UNLESS you move the file after queue to a new filename containing the Agent
Number.
like ;
exten = whatever,n,SET(MONITOR_FILENAME=blah-blah)
exten = whatever,n,Queue(${params}); Queue should contain option c to
continue in dialplan when callee hangup. Caller hangup case needs special
attention too
exten = whatever,n,System(mv ${old-Filename}
${old-Filename}-${MEMBERINTERFACE})
I guess this should do the job.
On Tue, Sep 27, 2011 at 8:30 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
Dears;
I am facing now a problem in the recording the calls that coming via the
queue, the problem that I am not able to make the filename contains the
agent (for example its extension) who received the call.
Actually by looking to the below settings, it is clear that the agent name
(it the phone extension or it is sip username .. etc) will not be included
in the filename.
How can I include the agent name in the filename? Because in outboud it is
easy as the ${CHANNEL} will contain the sip username of the IP Phone but in
the outbound it will contain the DAHDI channel that the call came via it ..
so How to inlude the sip username for the IP Phone of the agent that is
going to get the call from the queue?
exten =
s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = s,2,Queue(OrangeCMG,t,,,180)
exten = s,3,Macro(voicemail,SIP/reception)
Regards
Bilal
; If set to yes, just prior to the caller being bridged with a queue
member
; the following variables will be set
; MEMBERINTERFACE is the interface name (eg. Agent/1234)
; MEMBERNAME is the member name (eg. Joe Soap)
; MEMBERCALLS is the number of calls that interface has taken,
; MEMBERLASTCALL is the last time the member took a call.
; MEMBERPENALTY is the penalty of the member
; MEMBERDYNAMIC indicates if a member is dynamic or not
; MEMBERREALTIME indicates if a member is realtime or not
;
;setinterfacevar=no
Basically the variable ${MEMBERINTERFACE} will have the extension (if
using dynamic members) or the agent number.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Question about Registrations
One way of doing something when a peer registers is to use AMI to monitor events and when a register event occurs do what you want. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 23, 2011, at 7:08 PM, Alex Balashov wrote: On 09/23/2011 09:59 PM, CDR wrote: In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? No. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] Playback while dialing out
A(x) does not accomplish this. It completes the playback and then dials. What I would like is that dialing should start in parallel and playback should stop as soon as early media or ringing starts. Similarly, music-on-hold is not an option, it's too hard coded, I like to be able to change playback file dynamically. Any hints?? On Fri, Aug 19, 2011 at 7:14 AM, Eric Wieling ewiel...@nyigc.com wrote: Take a look at the A(x) and m options to dial. In the Asterisk CLI core show application dial for a the docs to Dial(). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Thursday, August 18, 2011 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Playback while dialing out Hi, please help me with dialplan below. My current dialplan looks like this, it plays a file and then connects the caller to my phone by dialing out. As you can see, it waits for file to be played completely before dialing out. What I would really like is that it should play the file (preferably repetitively) and simultaneously dial out the number, playback should stop as soon as dial answers or early media detected. exten = 500,1,Answer exten = 500,2,Playback(wait-while-we-connect-you) exten = 500,3,Dial(SIP/14085551234@myprovider) How do I make it work? -- _ -- 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 -- _ -- 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] Playback while dialing out
I am not sure you even read my mail, no music on hold option - it should work dynamically with any file. On Fri, Aug 19, 2011 at 6:18 PM, bakko asannu...@gmail.com wrote: Hi, you can configure a new music on hold, example: nano /etc/asterisk/musiconhold.conf [default1] mode=files directory=moh1 and put the audio file in this directory; then change your dialplan like: exten = 500,1,NoOp exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1)) exten = 503,3,Hangup 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 -- _ -- 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] Gtalk channel problem
We have been using gtalk channel from a long time now. It was working fine so far but from yesterday we are having problem. When gtalk destination is dialed and even answered, channel remains in Ringing state. Is there anything changed on google side? Jim -- _ -- 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] Playback while dialing out
Hi, please help me with dialplan below. My current dialplan looks like this, it plays a file and then connects the caller to my phone by dialing out. As you can see, it waits for file to be played completely before dialing out. What I would really like is that it should play the file (preferably repetitively) and simultaneously dial out the number, playback should stop as soon as dial answers or early media detected. exten = 500,1,Answer exten = 500,2,Playback(wait-while-we-connect-you) exten = 500,3,Dial(SIP/14085551234@myprovider) How do I make it work? Thanks -- _ -- 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 scaling
convert mp3 to sln, this itself will give you quiet a big capacity boost. On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen m...@bellcom.dk wrote: On 2011-08-16 21:14, Warren Selby wrote: Is it going to be just one mp3 stream that is shared across all users (I.e everyone hears the same thing at the same time), or is it 1000 separate mp3 streams (everyone always starts at the beginning of whatever they are hearing). It's a shared stream. When testing now, new listeners doesn't spawn new mpg123 processes. Are you going to have reliable timing generation on an EC2 instance, since IAX streams and music on hold playback will sound bad if the timing isn't good. We are using the zaptel and ztdummy kernel module, and we haven't noticed any problems with the audio quality yet. Should we be worried about this when the load gets higher? Will you have sufficient bandwidth allocated to you for that many simultaneous calls? Good point. We will have to do some calculation and research on what EC2 offers here. Is there going to be any codec transcoding going on? Can you generate your streams in the preferred codec, instead of mp3? The source is an icecast server streaming mp3. I haven't figured out a way to get around that. But from what I understand its just one reencoding for all the listeners. I think if you're just using one stream spread across all the callers, you'll have much better performance from the system as a whole. You may want to look at the quality differences between a SIP trunk and an IAX trunk as well. I had a talk with our IAX2 trunk provider and they told me that we could expect better performance from a SIP trunk. They also had a limit on 2000 channels, so we may have to look for another trunk. Are there any tools or services to simulate a lot of IAX2 or SIP users that you can recommend? How do you test how many users an asterisk system can handle? Thank you for taking the time to reply. Morten Thanks, --Warren Selby, dCAP On Aug 16, 2011, at 10:16 AM, Morten M. Hansen m...@bellcom.dk wrote: Hi I'm hoping someone could comment on how our setup will perform under larger loads. Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2 large instance (7GB RAM, 2 virtual cores with EC2 compute units). Using an IAX2 trunk we offer normal phones to dial in and listen to a mp3 stream using music on hold. If we wanted to let 1000 users listen to the stream at the same time, would that be possible? What limits will we hit? How about 1 users? Regards Morten -- _ -- 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 -- _ -- 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 -- _ -- 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] Problem setting for incoming termination
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
[dallas]
type=user
username=dallas
secret=somepassword
host=dynamic
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
accountcode=411
context=dallas
This is the configuration at client end.
[outgoing]
type=peer
username=dallas
secret=somepassword
host=ipaddress
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
We do not require the client to register, neither we want them to use
fromuser field. I think we are doing some silly mistake since this
should be a simple configuration used by many. Please help
Thanks
Jim
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Re: [asterisk-users] BT killed Ribbit
we too had enough issues with ribbit support and moved to tringme for web based phone. On Wed, Aug 10, 2011 at 3:29 PM, Dean Collins d...@cognation.net wrote: Any thoughts on why they did this? - http://venturebeat.com/2011/08/09/bt-kills-ribbits-web-phone-platform-sends-customers-to-the-fast-growing-twilio/ what makes Twillio successful but another company willing to kill off a $100m+ investment? Cheers, Dean -- _ -- 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 -- _ -- 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] Problem setting for incoming termination
Anyone?
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
[dallas]
type=user
username=dallas
secret=somepassword
host=dynamic
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
accountcode=411
context=dallas
This is the configuration at client end.
[outgoing]
type=peer
username=dallas
secret=somepassword
host=ipaddress
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
We do not require the client to register, neither we want them to use
fromuser field. I think we are doing some silly mistake since this
should be a simple configuration used by many. Please help
Thanks
Jim
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Re: [asterisk-users] Problem setting for incoming termination
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
[dallas]
type=user
username=dallas
secret=somepassword
host=dynamic
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
accountcode=411
context=dallas
This is the configuration at client end.
[outgoing]
type=peer
username=dallas
secret=somepassword
host=ipaddress
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
We do not require the client to register, neither we want them to use
fromuser field. I think we are doing some silly mistake since this
should be a simple configuration used by many. Please help
Thanks
Jim
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[asterisk-users] DAHDI Callerid and transfer problem
Hi, We need some help.
We are unable to transfer the incoming call from DAHDI to another
number. We are able to receive calls and dial out fine, but what we
really want to achieve is to transfer the call so that PRI will be
free and also the transferred number will get the received callerid.
Here is dialplan.
[default]
exten = 1999001,1,myprint(${CALLERID(num)})
exten = 1999001,n,transfer(DAHDI/1/14085551234)
exten = 1999001,n,myprint(${TRANSFERSTATUS})
The problem is transfer fails and TRANSFERSTATUS is set to
UNSUPPORTED. It works if we change 'transfer' command to 'Dial' but in
this case it does not pass callerid. facilityenable and transfer is
set to 'yes' in chan_dahdi.conf.
Any hints what we are doing wrong?
Thanks
Jim
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Re: [asterisk-users] DAHDI Callerid and transfer problem
Please help. I would be surprised if no one ever faced this problem
On Mon, Aug 8, 2011 at 3:16 PM, Jim Boykin boykin...@gmail.com wrote:
Hi, We need some help.
We are unable to transfer the incoming call from DAHDI to another
number. We are able to receive calls and dial out fine, but what we
really want to achieve is to transfer the call so that PRI will be
free and also the transferred number will get the received callerid.
Here is dialplan.
[default]
exten = 1999001,1,myprint(${CALLERID(num)})
exten = 1999001,n,transfer(DAHDI/1/14085551234)
exten = 1999001,n,myprint(${TRANSFERSTATUS})
The problem is transfer fails and TRANSFERSTATUS is set to
UNSUPPORTED. It works if we change 'transfer' command to 'Dial' but in
this case it does not pass callerid. facilityenable and transfer is
set to 'yes' in chan_dahdi.conf.
Any hints what we are doing wrong?
Thanks
Jim
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Re: [asterisk-users] Connecting to a Taqua switch
My provider has always sent the SIP control info from one IP and the media packets from another. As long as your firewall passes the data there should be no problem. I did not have to do anything special in my configuration. This is using ABE which is based on 1.4. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 22, 2011, at 10:09 AM, Philip Prindeville wrote: Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN via SIP on a Taqua 7000 switch? My local carrier recently upgraded software and changed their configs so that signalling and media are on different cards (and hence different IP addresses), and it's causing issues. I suspect there are other factors at play... it may or may not be behind a properly configured SBC. Thanks, -Philip -- _ -- 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 -- _ -- 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] DTMF issues still
I had a very strange problem with a Sangoma card that I had both Sangoma (about 3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma tech to look at the problem it went away. I told the tech he did something and he said I alway verify the firmware on the card is updated and as it was not I updated it. That fixed the problem. This system had worked before a dahdi update was applied. Bottom line make sure you have the most current firmware for your card. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote: I am still having major issues with dtmf recognition. My setup is Polycom end points. Tried this with different models, firmware and cfgs. Outbound calls are not going out reliably. Phones are set to rfc2833. I have had sangoma and elastix support look at it.. No better. Running asterisk 1.8.4. What am I missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on all lines no problem. Sangoma card is a a400 with echo cancel. Sent from my android device. -- _ -- 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 -- _ -- 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] Agent Login, Logout, Ready, Not Ready from the CTI application
You need to use the AMI interface an deal with the events that are give to you. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote: Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an application and to be integrated with the CRM, how to acheive this? Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which is embded in the CRM application) will receive the the caller id or information via that CTI client. How this to be done in Asterisk? By the way: is the ready and not ready in Asterisk? Regards Bilal -- _ -- 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 -- _ -- 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] chanspy spies on wrong channel
The argument to chanspy is a pattern and not an exact match. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 2, 2011, at 3:48 PM, steve casto wrote: asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel 2.0 (from extensions.conf) exten= 304,1,ChanSpy(Zap/4|q) exten= 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304@flash:1] ChanSpy(Zap/31-1, Zap/4|q) in new stack [Jul 1 18:24:47] VERBOSE[14447] logger.c: == Spying on channel Zap/41-1 [Jul 1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to Zap/41-1 If while spying on Zap/41 that channel is hung up: [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel Zap/41-1 [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1 [Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to Zap/4-1 thanks list Steve -- _ -- 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 -- _ -- 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] background audio for inbound leg
The way I play a sound file into a bridged call is to use chanspy w option. I do this with an application that does AMI commands. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 17, 2011, at 10:25 AM, Tom Browning wrote: Is there an easy way to feed an audio file (think background music, ever so softly) to the inbound leg of a bridged call (and not send / mix it to the outbound leg)? exten = blah,1,Answer() exten = blah,2,StartSomeAudio(foo)? exten = blah,3,Dial(SIP/bar) Where the audio would continue to play to the inbound leg in addtion to the bridged inbound audio. Thanks in advance including any RTFM references :-) Tom -- _ -- 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 -- _ -- 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] issues.asterisk.org/jira not working
I get this on my Mac: Safari can’t open the page. Safari can’t open the page “https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t establish a secure connection to the server “issues.asterisk.org”. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 11:38 AM, William Stillwell wrote: You mean this one? https://issues.asterisk.org/jira/browse/ASTERISK-17984 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 2:17 PM To: asterisk-users Subject: [asterisk-users] issues.asterisk.org/jira not working Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- 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 -- _ -- 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] issues.asterisk.org/jira not working
If I click on the link below, without jira, Safari goes to here: https://issues.asterisk.org/main_page.php And yes it works. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 1:54 PM, Kevin P. Fleming wrote: On 06/08/2011 02:27 PM, Andrew Latham wrote: On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryantruss...@digium.com wrote: A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is SSLCertificateChainFile /full/path/to/your.ca-bundle Can Safari open a connection to https://issues.asterisk.org? (no /jira suffix) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- _ -- 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] Trying out a new version with sangoma card
In asterisk CLI do pri show spans. The fact the card is in RED alert means the hardware does not see the pri line connected to the card. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 11, 2011, at 6:55 PM, Nicolas Ross wrote: Le 2011-05-09 09:31, Jim Dickenson a écrit : Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. It appears it did not change anything... So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, on the asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1. When asterisk is running, cat /proc/dahdi/1 yields : Span 1: WPT1/0 wanpipe1 card 0 (MASTER) B8ZS/ESF RED 1 WPT1/0/1 Clear (In use) 2 WPT1/0/2 Clear (In use) (...) 24 WPT1/0/24 Hardware-assisted HDLC (In use) And when it's not, the (In use) go away. When, dialing I get Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) So, does anybody got any idea ? -- _ -- 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 -- _ -- 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] Trying out a new version with sangoma card
Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 9, 2011, at 6:11 AM, Nicolas Ross wrote: Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via PPP. Now, I want to freshen this setup to something newer. So I installed a Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers and an A101 card I had laying around. I did a test this weekend and pluged in our PRI in that test server. I never got succeded to have a call trough. When I dialed in, the call is hanged up with : Channel 1/1, span 1 got hanup, cause 6 Spawn extension (ael-default, s, 3) exited non-zero on 'DAHDI/i1/NPANXX-2' Hungup 'DAHDI/i1/NPANXX-2' Here's my dahdi/system.conf : loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:0 bus:6 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 echocanceller=mg2,1-23 hardhdlc=24 my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with : switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 as the last non-commented lines. So, for one thing, the card I have in my test server doesn't have an hardware echo canceller, but it's still enabled in my wanpip setting. Could that be a source of problem ? Other than that, is there anything obvious I've missed ? -- _ -- 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 -- _ -- 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] HA Asterisk
On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal Correct, there are no cards inside any system. You have an external box that can have a combination of PRI, FXO and FXS ports; depending on need. The external box is connected via USB to the two systems. The twinstar option allows you to connect the external box to two systems via USB and provides fall over from primary to secondary on failure of the primary.-- _ -- 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] Orginate not working well with PSTN lines
Originate successfully queued only means that the originate action was handed off to asterisk not that is was executed yet. There are other events, depending on which events you are reading, that tell you the call was answered and such. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote: Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890 Response: Success ActionID: ABC45678901234567890 Message: Originate successfully queued -- Remote UNIX connection disconnected Channel DAHDI/1-1 was answered. -- Executing [1234@outbound-ivr:1] SayDigits(DAHDI/1-1, 1234) in new stack -- DAHDI/1-1 Playing 'digits/1.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/2.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/3.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/4.gsm' (language 'en') -- Executing [1234@outbound-ivr:2] Playback(DAHDI/1-1, demo-congrats) in new stack -- DAHDI/1-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [1234@outbound-ivr:3] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' Thanks Regards, Ashik -- _ -- 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 -- _ -- 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] DTMF not being sent ( RFC2833 )
I had problems with a system I was trying to bring up using a couple older a104d cards we had lying around. Neither card would pass audio. I worked with one Sangoma tech for a couple hours while he tried various things. The second tech I worked with got on the system and updated the firmware for the cards. When I tried to show him the problem things worked. I said you did something as this did not work an hour ago. He told me the first think he does when troubleshooting is to update the firmware to the current version. A lesson I have now learned. I do that with software but rarely remember to look for firmware updates. Take a look at wiki.sangoma.com and it lets you know current firmware versions as well as how to update if you are not running the current version. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 25, 2011, at 4:41 PM, Edwin Lam wrote: i think i have similar problem after upgraded from 1.4.x to 1.6.2.17. (originally upgraded to 1.8.3.2 unfortunately there were other more pressing problems that forced me to downgraded it to 1.6.2.17) i have a wanpipe device with 2 channels uses PRI signalling to PSTN the other 2 uses FXO signalling (connect to Rhino FXS channel bank). the PRI part works fine but the FXO channels are having DTMF digits skipped. i'm still trying to find out what's wrong with it. On 4/23/11 8:48 AM, David wrote: Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 mailto:SIP/114186939...@pri1.omnity.net,30,D(132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk 1.6 current with centos ( os that asterisk is developed on from my understanding ) with all default settings ( excluding logger.conf, dialplan and outboundproxy ) and I am having problems with the DTMF. Both servers were installed with CentOS 5.5 and were updated last night, after which I reinstalled asterisk. This did not resolve the issue. I am at wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a 503.
Has anyone else noticed this? I went through the issue tracker but I couldn't
find any relevant bug posted in the past. I am certain that in previous
versions I could set the reply message to the desired value, so I wonder if
this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It is
that the Hangup(cause) command seems to ignore its argument and just sends
a 503 cause to the caller for all unanswered calls no matter what...
Hangup(cause) was working as expected in previous versions and I wonder if
something was broken along the way that went by unnoticed. I am just asking
in the list in case I am missing something too obvious before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
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Re: [asterisk-users] Reach PSTN from another Asterisk
On server B use IAX2/iax-trunk-name/whatever-needs-to-be-dialed-on-A-to-make-call in the Dial command. like Dial(IAX2/sfserver1/9411212) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 15, 2011, at 12:29 PM, Alejandro Cabrera Obed wrote: Dear, we have the following: - Asterisk A with E1 to PSTN connection. - Asterisk B with IAX trunk to Asterisk A - Outgoing routes between Asterisk A and B - Asterisk A with an outgoing route to PSTN with 9|. dial rule How can I reach the PSTN from Asterisk B through Asterisk A ??? Thanks a lot !!! Alejandro -- _ -- 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 -- _ -- 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] Microsoft Lync server and Asterisk access
We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to connect to an Asterisk box? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
Do you have the Sangoma wanpipe software installed? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 13, 2011, at 7:37 AM, satish patel wrote: Try dmesg command root@:~# dmesg | grep -i Sangoma [ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma Technologies Inc. From: kaushalshri...@gmail.com Date: Wed, 13 Apr 2011 19:38:06 +0530 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card Hi, I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card installed on the box. Its not detected. Details are as below :- [root@asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller 00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller 00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10) 00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller 00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge 00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 Audio Controller (rev 01) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 200G Series] 01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480) 02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI Bridge (rev aa) 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 20) 05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet Adapter (rev 11) [root@asterisk ~]# cat /etc/redhat-release CentOS release 5.5 (Final) [root@asterisk ~]# asterisk -v Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. [root@asterisk ~]# Please suggest/guide Thanks Kaushal -- _ -- 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 -- _ -- 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 -- _ -- 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] send voicemail to multiple emails
If you want externnotify to not fire when someone checks then put in a new option in voicemail.conf to have it work that way. Then contribute that change and it might be accepted. externnotify_on_check: yes|no or some such thing. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 12, 2011, at 1:52 PM, Steve Edwards wrote: On Tue, 12 Apr 2011, vip killa wrote: Honestly, I don't understand why externnotify should run when someone checks their voicemail... the change i made, makes sense so maybe that should be contributed to the asterisk source. Even if it makes sense to everybody on the list, changes that conflict with documented and implemented behavior that other users may be depending on are unlikely to be accepted. -- 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 -- _ -- 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] Variable inheritance with dialplan command Originate
Another option is to pass the information in the extension. At times I have an
extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1 and
info2 values. Not real pretty but as this is computer generated calls it gets
the job done.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 8, 2011, at 8:57 AM, Naomi Rosenberg wrote:
Thanks. That's as I thought (feared). Dial is not an option in this case but
I have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, 8 April, 2011 4:35:43 PM
Subject: Re: [asterisk-users] Variable inheritance with dialplan command
Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the
Originate() dialplan command, variables prefixed with two underscores
would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018,
__myvar=foo) in new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2,
) in new stack
-- Executing [123@test_orig:2]
Hangup(Local/123@test_orig-cbab;2, ) in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you
expected.
-- Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] AMI redirect from Queue to MeetMe
I would be surprised that you did not always hang up the second channel you are
redirecting. Once you transfer one leg there is nothing connected to the second
leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then setup a
call to the agent and meetme room.
More or less like:
Action: Redirect
Channel: SIP/GXP280_18-0001
Exten: do_meetme601MyID
Context: cfmc_cdi_private
Priority: 1
ActionID: MeetMe
Async: true
Action: Originate
Channel: Agent/1001
Exten: do_meetme601MyID2
Context: cfmc_cdi_private
Priority: 1
ActionID: DirectMeet
Async: true
exten = _do_meetme.,1,UserEvent(BeforeMeetMe,Info:${EXTEN:9} ${UNIQUEID}
${CHANNEL})
exten = _do_meetme.,n,Answer()
exten = _do_meetme.,n,Set(CfMC_RoomToUse=${EXTEN:9:3})
exten = _do_meetme.,n,Set(CfMC_CurrentID=${EXTEN:12})
exten = _do_meetme.,n,Set(MEETME_MOH_CLASS=meetme-music)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten = _do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011, at 1:23 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
I have scenario as follows,
A call comes to queue.
Available agent will answer the call.
BridgeEvent wil be generated in AMI with channel1 and channel2.
Parse channel1 and channel two from the event and redirect them to a meetme
room,
Dialplan,
Exten = 1234,1,MeetMe(1234,1dq)
But sometime it works and sometime one leg gets disconnected after
redirection. Is it a bug to asterisk-1.6.2.13 ?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
Important notice: This e-mail and any attachment there to contains corporate
proprietary information. If you have received it by mistake, please notify us
immediately by reply e-mail and delete this e-mail and its attachments from
your system.
Thank You.
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Re: [asterisk-users] Executing shell commands via AMI
If you want total control from AMI then point at an extension that you can set variables to commands and arguments, call an AGI and set variables that can be passed back to AMI via user events. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 16, 2011, at 3:03 PM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: Wednesday, March 16, 2011 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Executing shell commands via AMI But what about if asterisk running with non-privilege user? Still it is not a good idea. Yes I forgot to say that I also run Asterisk as a regular user, which also helps with security. But I really don't see much of a threat on this. AGI does almost the same. This won’t help but I’ll chip in anyway. In AGI, you have “total” local control. In AMI, it’s a crap shoot. -- _ -- 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 -- _ -- 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 do you handle queues with AMI?
What we do is just before the call to queue we do a userevent that has the uniqueid and the channel and any other information we care about. You can hold on to this information and match it when you get the agentconnect event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 11, 2011, at 7:21 AM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 11, 2011 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How do you handle queues with AMI? Hey all, I’m in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I’m writing this in C# but language isn’t important. I’m not looking for a hand out on code, what I’m really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I’ve already got the app to recognize the “packets” of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I’m missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I’m writing this, if anyone would like to help out or share code I’m up for it. I’ll make my code available to all interested in doing this in C# (it’s pretty painless). Thanks! Louis If you look through your CDR, you’ll see the information you need to develop this methodology. Keep in my that (as I understand it), when an agent picks up a call, the uniqueid will change just like the call had been transferred. -- _ -- 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 -- _ -- 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] ChanSpy with alphanumeric SIP channels [1.6.2]
I think in the chanspy application you can give it a template to prepend to
what is entered. If you do chanspy(ab_) you might be able to enter the
remaining digits.
Short of that you can set up a loop that reads the digits, calls
chanspy(ab_${digits}), if the version you are using has my S option then * will
exit the chanspy app and you can loop back to the top.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 9, 2011, at 6:28 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits). ChanSpy is working fine for listening in to conversations
initiated by these channels, and I can use '*' to randomly switch
channels. However, is there any way in this scenario to be able to
switch ChanSpy to a specific channel by typing in a ...# key sequence
during a spy session?
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org/
GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F
PsyTrance Chill: http://schizoid.in/ || It is the mind that moves
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[asterisk-users] Connecting to Cisco Iad2430 to Asterisk
Is it possible to SIP trunk to this Cisco device so that phones connected to the Cisco box can dial extensions on the Asterisk box? What I would like to be able to do is dial some extension(s) on phones connected to the Cisco box and have the call routed into extension(s) on the Asterisk box. One of our clients has a call center with 65 analog phones connected to the Cisco box. We would like to be able add our dialer appliance into their operation without having to replace any more equipment than needed. We need an easy way for the agents to connect to an extension on our appliance that basically does an agentlogin. Ideas as to how to best accomplish this would be appreciated. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Determine When Call Is Picked Up In Queue
You might be able to use a macro on the dial command (option M) which gets run when the remote end answers. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 29, 2011, at 10:30 AM, Joseph Begumisa wrote: Hi, I have a situation where a call comes in to my asterisk server, goes through an IVR and is then handed off to another asterisk server where it enters a queue waiting for an agent to answer the call. (I do not control the second asterisk server). Is there a way for me to know when the call is actually picked up on the second asterisk server? I have a billing application that needs to start billing when the call is actually answered by an agent. Thanks a lot. Best Regards, Joseph -- _ -- 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 -- _ -- 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 check a number online or offline
You can always place a call to an extension that sends a user event from AMI. If there are no native AMI commands that can return what you want originate a call to a local extension that returns a user event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote: Thanks Dhaval, My purpose is that i want to use java application (using Asterisk Manager Interface) to check a number online, offline or unreachable. Your suggest uses function DEVICE_STATE but this is written in dialplan not application java. Do you know other way to do this for me?thanks and looks forward to listening your reply. Regards! Phuong On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards Dhaval On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.net wrote: On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Not really. It's a list of manager commands. There is 'SIPshowpeer' which will work for sip stuff. Try the command 'Command' action and you can send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work in some cases.. S -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Call Back on Busy
It should not be too hard to write some dialplan code that detects the busy, plays a sound file asking if you want to camp-on to the called device, read an answer and loop around checking device status and placing a call when both the calling device and called device are free. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 10, 2011, at 8:39 AM, John Novack wrote: That function in the telephony world is called camp-on Can't say for sure if Asterisk can do that, not which version, nor freepbx John Novack Ron wrote: Hi All, One of our user asked the question, when she tries to call another local extension but the other end is engaged she will keep on trying until she finally can get thru. So she asked would it be possible to request for an auto-callback from the user she's trying to call to once it's not engaged anymore. is this possible on asterisk? what is that feature called? i am using asterisk 1.4 with freepbx. Thanks in advance. Regards 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-users -- Dog is my Co-pilot -- _ -- 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 -- _ -- 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 check a number online or offline
If you do an AMI packet like this:
Action: Originate
Channel: Local/get_i...@some_context
Exten: do_noop
Context: some_context
Priority: 1
ActionID: GetInfo
Async: true
and then have a couple extensions that do what you want. Here is what I do in
my case:
exten = get_info,1,Answer()
exten = get_info,n,UserEvent(GetInfo,Version:ABE
DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)} CfMC:83351)
exten = get_info,n,Hangup()
exten = do_noop,1,Answer()
exten = do_noop,n,Wait(1)
exten = do_noop,n,Hangup()
You would then do what you need to do in your extensions.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:
Thanks Jim,
Can you say about your idea clearlier? I want to use AMI in an application
java to check a number online, offline or unreachable and result is returned
to the appliction java. If the number is online now, i will use AMI to hangup
it, else i do nothing.
Best regards,
Phuong.
On Mon, Jan 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:
You can always place a call to an extension that sends a user event from
AMI. If there are no native AMI commands that can return what you want
originate a call to a local extension that returns a user event.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:
Thanks Dhaval,
My purpose is that i want to use java application (using Asterisk Manager
Interface) to check a number online, offline or unreachable. Your suggest
uses function DEVICE_STATE but this is written in dialplan not application
java. Do you know other way to do this for me?thanks and looks forward to
listening your reply.
Regards!
Phuong
On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:
Hello ,
You can use Dialplan function DEVICE_STATE, which will gives you perfect
status of DEVICE.
regards
Dhaval
On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.net
wrote:
On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to resolve
this. Can you say clearlier for me?
Not really. It's a list of manager commands. There is 'SIPshowpeer' which
will work for sip stuff. Try the command 'Command' action and you can send
any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
in some cases..
S
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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that was not answered and I did not see any more information. The dumpchan of DADHI/23-1 did not happen as that is in a macro that only gets called for an answered call. I only see this: Executing [91112223...@empl:8] Dial(SIP/mine-0521, Dahdi/G1/111222|60|gM(out-dial)) in new stack DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 -- Requested transfer capability: 0x00 - SPEECH -- Called G1/111222 DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/23 span 1 -- DAHDI/23-1 is proceeding passing it to SIP/mine-0521 -- DAHDI/23-1 is ringing DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on -- DAHDI/23-1 answered SIP/mine-0521 -- Executing [...@macro-out-dial:1] DumpChan(DAHDI/23-1, ) in new stack Dumping Info For Channel: DAHDI/23-1: Info: Name= DAHDI/23-1 Type= DAHDI UniqueID= sys.domain.com-1294514614.2630 CallerID= 9111222 CallerIDName= (N/A) DNIDDigits= (N/A) RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x4 (ulaw) WriteFormat=0x4 (ulaw) ReadFormat= 0x4 (ulaw) 1stFileDescriptor= 35 Framesin= 189 Framesout= 176 TimetoHangup= 0 ElapsedTime=0h0m4s Context=macro-out-dial Extension= s Priority= 1 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: MACRO_DEPTH=1 MACRO_PRIORITY=1 MACRO_CONTEXT=from-outside MACRO_EXTEN= DIALEDPEERNUMBER=G1/111222 TRANSFERCAPABILITY=SPEECH DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0 DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on DAHDI/23-1 DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on DAHDI/23-1 -- Hungup 'DAHDI/23-1' == Spawn extension (empl, 9111222, 8) exited non-zero on 'SIP/mine-0521' -- Executing [...@empl:1] Verbose(SIP/mine-0521, 2|Hangup SIP/mine-0521 with cause 16) in new stack == Hangup SIP/mine-0521 with cause 16 -- Executing [...@empl:2] DumpChan(SIP/mine-0521, ) in new stack Dumping Info For Channel: SIP/mine-0521: Info: Name= SIP/mine-0521 Type= SIP UniqueID= sys.domain.com-1294514614.2629 CallerID= 444555 CallerIDName= Jim Dickenson DNIDDigits= 9111222 RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x2 (gsm) WriteFormat=0x2 (gsm) ReadFormat= 0x2 (gsm) 1stFileDescriptor= 65 Framesin= 248 Framesout= 253 TimetoHangup= 0 ElapsedTime=0h0m0s Context=empl Extension= h Priority= 2 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: DIALSTATUS=ANSWER DIALEDTIME=5 ANSWEREDTIME=1 RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00 BRIDGEPEER=DAHDI/23-1 DIALEDPEERNUMBER=G1/111222 DIALEDPEERNAME=DAHDI/23-1 MACRO_DEPTH=0 RCStatus=0 MyChan=SIP sipcallid=0b69233cd5469...@192.168.0.16 SIPUSERAGENT=Grandstream GXP2000 1.2.2.6 SIPDOMAIN=sys.domain.com SIPURI=sip:m...@00.00.000.000:5064 -- Executing [...@empl:3] ExecIf(SIP/mine-0521, 0|Set|DB(conf//haveadmin)=no) in new stack -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 7, 2011, at 12:44 PM, C F wrote: PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine
[asterisk-users] Benefit of PRI vs SIP trunk calls
Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine? If a call is placed to a number that is disconnected so a special information tone is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Moving asterisk from one network to another.
If you set bindaddr in any conf file you will need to change the IP address there. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 24, 2010, at 4:30 AM, Asterisk Man wrote: Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? It is assumed that DB is on the same box. Asterisk box has got Asterisk running in it with no issues. Probably, it should not complain. I tried to check for IP address in Asterisk files (using ‘find . | xargs grep 192.168.X.XX –sl’), but it seems that Asterisk does not store specific IP in file(s). Your thoughts on this if I m missing something. -AsteriskMan -- _ -- 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 -- _ -- 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] DIALSTATUS on CANCEL
If on the dial command you add option g, if the call is not answered, it will
fall through to the next statement which can be a hangup command and then it
will go to the h extension. If that does not then make the statement after the
dial command a goto h extension.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 24, 2010, at 6:03 AM, brya...@zktech.com wrote:
If a call is hung up before an answer our h extension is not running in our
dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] Asterisk SIP attacks and sshguard
I do not have log examples to provide but do have info about other issues. There is a nocolor option in asterisk.conf that can turn off color. logger.conf has a provision to use syslog directly. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 9, 2010, at 5:57 AM, Joe Greco wrote: Hello, We had been seeing SIP-guessing attacks on our Asterisk server here. While it wasn't that hard to write a once-a-minute cron job to spank the lusers, that runs once a minute and creates little spikes in the usage and I/O graphs, and is slower to respond than I'd really prefer. I felt that it'd be much cooler to get something more comprehensive put together. We don't use fail2ban because I don't like having to install python. sshguard is a high-performance compiled C application that can run off a log file or a pipe from syslogd to sshguard, meaning that it can respond a lot more quickly than once a minute, and works with very modest overhead on the host system. It also has features such as touchiness, so that it can get tougher on a miscreant as time goes on; my own shell script is naive in that once it passes a threshold, there's just a permanent rule generated. This worries me if I ever have a situation where a legitimate remote client gets messed up and tries the wrong password or something like that; sshguard does a much nicer job in this regard. In any case, my initial attempts to create rules for sshguard didn't work right, quite possibly because I don't often work in LEX/YACC. I submitted a request to the sshguard guys suggesting new rules. http://www.sshguard.net/support/submission/detail/49ce7182028d8b6f3e3d/ and on their mailing list, a little more: http://sourceforge.net/mailarchive/forum.php?thread_name=F4E10075-5D93-43B4-B73A-1FD217BE130D%40sshguard.netforum_name=sshguard-users In particular, they're looking for log examples of some of those messages, but I have no idea how to generate the conditions that would cause these messages. I'm also not sure if there's a way to disable color codes in the Asterisk log files; we log indirectly via BSD's logger # asterisk -vvv 21 | logger -t asterisk so it may be thinking that the console is color-capable. We use this method because this forces them through the syslog mechanism; we need that for centralized logging, and it's handy for things like sshguard too. Specifically looking for examples of (or how to generate) 1).*No registration for peer '.*' (from HOST) 2).*Host HOST failed MD5 authentication for '.*' (.*) 3).*Failed to authenticate user .*@HOST.* If anyone who is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. Thanks. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- 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 -- _ -- 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] What are the minimal permissions required to read the PeerStatus and Registry events?
A problem that I have always had with AMI events is that they are not controllable at a very fine level. As an example, turning on call class gives WAY more that one might always want. I had posted a patch some time ago that added a new class. The patch was rejected with a comment that some work would be undertaken for 1.8 that would allow finer control. As I am using the ABE version of asterisk I have not looked at 1.8 to see if anything has been done. It would be very useful if there was more control over which events one sees. As I recall all the events have a name and if one could somehow say I want to see events with names x,y,z and forget the whole class mess. That way if you want to see some specific events but not others you could get what you want. I know this does not answer you question but I for sure feel your pain. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 3, 2010, at 10:36 AM, Frank Church wrote: I am logging events from the AMI and the PeerStatus and Registry events show that the privilege for them is System,All. Can a lower set of privileges be used? All looks pretty high to me. /Frank -- _ -- 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 -- _ -- 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 hangup all channels
Can also do asterisk -r -x 'restart now'
asterisk*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 27, 2010, at 8:45 AM, Steve Edwards wrote:
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Installing Asterisk to it's own directory
What you did is what I would have done. That way the executables have their conf file location adjusted and everything will be inside the specified --prefix location. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 20, 2010, at 5:48 AM, Stephen Brown wrote: Thanks... I actually did a ./configure --prefix=/root/asterisk18 and ended up with this: -- _ -- 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
