[asterisk-users] AMI: Local Channels
Hello, I'm using Asterisk 1.6.2.10. Whenever I dial Local channels via asterisk manager, the calls never get a hangup signal even with timeout specified. I find channels with ZOMBIE text appended. It ends up occupying all the channels with the result that asterisk thinks every channel is busy, hence drops further calls. Also, all calls dialed out through the local channel, get the cdr populated before hangup (obviously with incorrect information). If someone else has gone through this problem please share and let me know how to rectify the issue. -- 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 auto-dial out a SIP .
thnks for the reply.. i want to know is there any way to call a SIP to SIP by command line regards Upendra -- _ -- 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 auto-dial out a SIP .
On Thursday 01 March 2012, upendra wrote: thnks for the reply.. i want to know is there any way to call a SIP to SIP by command line Yes. Just write a script in your favourite language (even bash will do if there is nothing better) to set up a callfile, then invoke it from the command line. As this list forbids discussion of paid-for services, you will need to contact me separately if you want me to write the script for you. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line noise/hiss on Openvox A400P card on FXO
I have a server with an OpenVox A400P card with 2 FXO modules on it. The internal extensions are SIP Grandstream phones. When making or receiving external calls through PSTN, there is an interrupted hissing like high pitch noise - which might go away for few seconds then start again. 1. The noise is not present when calling in between internal extensions (SIP only). 2. The noise is the same on both PSTN lines. 3. The noise is NOT present when I tried two different phones directly in the PSTN line(s) (a Philips DECT phone and a BT Converse phone) Is the noise interference actually on the line, which the phones filter out because of their better electronic design (then the OpenVox card) - or is it generated somewhere in the server or on the OpenVox card? I have tried: 1. Checking the interrupts and making sure the OpenVox card has its own IRQ. 2. Moving the card around on different PCI slots. 3. Changing the second network card with a different model (the first one is integrated in the motherboard). 4. Changing the motherboard, CPU and RAM (one motherboard AMD with Sis chipset, the other one Intel). 5. Placing ferrite cores on the phone cables. 6. Checking to see if the OpenVox card gets 1000 interrupts per second and it does. 7. Upgrading the kernel from 2.6.29 to 2.6.37 8. Ran FXO tune and made sure it starts with DAHDI 9. Disabled and enable software echo cancellation - it makes no difference. The server is virtually under no load during the tests. It does have IDE hard-drives (which apparently can cause problems) - but there is not much I can do about that. I also have a Sangoma USB FXO adapter - which I'm about to install and configure to see if it makes a difference. I would really like to figure out where is the noise coming from - as I'm going a bit in circles. If I can find out for sure that the OpenVox card is either broken or low quality - I'll just have to replace it. But I can't even figure that out for sure. The specs are: CPU: Celeron 2.4GHz Asterisk 10.1.2 Dahdi 2.6.0 Hard-drives: IDE OpenVox A400P analog card Many thanks for any advice. -- _ -- 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] SS7 Disposition
In almost all major releases of asterisk 1.6.x, SS7 Disposition never sets to ANSWERED, even when someone answers the call, it logs NO ANSWER in the cdrs. Please help me resolve the issue. -- 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] SS7 Disposition
Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] SS7 Disposition
What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] SS7 Disposition
Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.comwrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] SS7 Disposition
Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Date: Thu, 1 Mar 2012 18:23:47 To: vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.comwrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] asterisk distributions
Howdy, I have tried all of these and a few more. PBXinaFlash gave me the best results, by far. AsteriskNow produced a basic working system. I could not get any of the others configured to work at all. I should tell you my restrictions. I was evaluating these distros to see which one I could use to teach at a local computer group. I wanted to do very little configuration through the command line, since my goal was not just to get a working system, but to have something I could easily show others how to setup. And, I was using real phone hardware. My phone and line were driven from a Digium TDM400. The AsteriskNow system only worked because someone on IRC helped me find a couple of obscure setting, but it does work. So, it somewhat depends on your needs, but I'd go with PBXinaFlash. And, I added the IncrediblePBX package. It is not perfect. I am now trying to add IAX trunks, and the mysteries involved make that slower than I would like. Good luck -- _ -- 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] SS7 Disposition
I tried it on asterisk 1.8, and it worked fine. On Thu, Mar 1, 2012 at 6:39 PM, Vinod Dharashive vdharash...@gmail.comwrote: ** Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -- *From: * [Digital^Dude] ® millennium@gmail.com *Date: *Thu, 1 Mar 2012 18:23:47 +0500 *To: *vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject: *Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.com wrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] outbound fax over t38 gateway can't pass
On 02/29/2012 02:28 PM, Dmitry Melekhov wrote: btw, played with res_fax.conf if I set maxrate=7200 fax machines try (and fail) 9600 anyway. Why? If limited ti 7200? looks like bug... Why do you think everything you don't understand is a bug? What you see is correct behaviour. Any party in the FAX chain can block V.17, or V.17+V.29. Only the entity sending a FAX can block individual modes. That's just how the FAX protocol works. So I set maxrate=4800 and modems=v27. Faxes pass Looks like problems with V29... I told you before what where the problem lies. It won't change by posting more messages like this. 29.02.2012 07:56, Dmitry Melekhov пишет: Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500- asterisk- ooh323- cisco 3845- fax machine fax can't pass. always reproducable. as I see in tcpdump produced dump fax machines tries to connect on 9600 and failed, no attempt to down speed. If I send fax in path panasonic 500 extension fax machine - asterisk (ReceiveFAX) it is received successefully all the time. If I send fax from asterisk with SendFax as following: asterisk(SendFax) - panasonic500-asterisk- ooh323- cisco 3845...- fax machine it always passes. Usually on 7200, sometimes on 4800. So ooh323 works OK, fax part works OK, t38 works OK, but not with fax machine (we tested to different). Inbound faxes in reverse direction, i.e. fax machine...cisco3845- ooh323 - asterisk - panasonic - fax machine always pass on 7200. More info is here https://issues.asterisk.org/jira/browse/ASTERISK-19436 Bug report was closed because not a bug :-) Could you help me solve this problem? Thank you! 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
Re: [asterisk-users] asterisk distributions
On Thu, 1 Mar 2012, Ralph Green wrote: Howdy, I have tried all of these and a few more. PBXinaFlash gave me the best results, by far. AsteriskNow produced a basic working system. I could not get any of the others configured to work at all. I should tell you my restrictions. I was evaluating these distros to see which one I could use to teach at a local computer group. I wanted to do very little configuration through the command line, since my goal was not just to get a working system, but to have something I could easily show others how to setup. And, I was using real phone hardware. My phone and line were driven from a Digium TDM400. The AsteriskNow system only worked because someone on IRC helped me find a couple of obscure setting, but it does work. So, it somewhat depends on your needs, but I'd go with PBXinaFlash. And, I added the IncrediblePBX package. It is not perfect. I am now trying to add IAX trunks, and the mysteries involved make that slower than I would like. Good luck I too tried PIAF and while it worked, the big problem I had with it and the reason I dumped it was because a lot of the scripts are compiled and encrypted. This restricts what you can do with the system without reinventing what they have already done. It is also possible this has changed as I have not looked at PIAF in a couple of years. PIAF is also very attractive because of the addons provided by Nerd Vittles and company. Some of them bolt onto asteriskNow without much difficulty others not so easily. I settled on AsteriskNow and have had it running for a couple of years. It really just depends on what you want to do with the system. If you do not mind the closed nature of the PIAF custom scripts than that can be a good choice. My next adventure is going to be testing the freepbx distro. If looks like it should be easy to get going and support but I have not tried it. Regards, -- Tom m...@tdiehl.org Spamtrap address me...@tdiehl.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk distributions
Tom you're killing me with the me's please! -- _ -- 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 distributions
On Thu, Mar 1, 2012 at 10:35 AM, Nick Khamis sym...@gmail.com wrote: Tom you're killing me with the me's please! -- hahahaha!! I've tried Elastix and FreePBX, for almost 4 years. Both are excellent!! _ -- 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] SS7 Disposition
Hi , I am using asterisk 1.6.1, any idea patch for the same Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Date: Thu, 1 Mar 2012 19:58:13 To: vdharash...@gmail.com Cc: asterisk-userasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SS7 Disposition I tried it on asterisk 1.8, and it worked fine. On Thu, Mar 1, 2012 at 6:39 PM, Vinod Dharashive vdharash...@gmail.comwrote: ** Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -- *From: * [Digital^Dude] ® millennium@gmail.com *Date: *Thu, 1 Mar 2012 18:23:47 +0500 *To: *vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject: *Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.com wrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- 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] asterisk distributions
On Thursday 01 March 2012, Ralph Green wrote: Howdy, I have tried all of these and a few more. PBXinaFlash gave me the best results, by far. AsteriskNow produced a basic working system. I could not get any of the others configured to work at all. I should tell you my restrictions. I was evaluating these distros to see which one I could use to teach at a local computer group. I wanted to do very little configuration through the command line, And *that* is where you were going wrong. Look, the command line is a fact of life. Microsoft have spent a fortune telling you that you're not smart enough to use it. You do not have to fall for that. Are you going to sit back and let them call you stupid? Think of trying to make yourself understood in a foreign country by pointing and gesturing. There comes a point where you will actually have expended *more* effort than if you had just bitten the bullet and learned the language in the first place. since my goal was not just to get a working system, but to have something I could easily show others how to setup. Again, this is where the command line excels. Irrespective of how the user of the computer has set up the GUI -- what icon theme they have selected or how they have arranged the menus, which GUI tools are present and so forth -- the command line method will always be the same. You really aren't doing your students any favours if you are teaching them blindly to avoid what is basically the most powerful feature of a GNU/Linux system. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using AMI and Telnet to place calls
Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 No Circuit Available back from xxx.xxx.xxx.xxx:5060 [Mar 1 12:55:50] Channel SIP/provider was never answered and be able to relate that back to the dialed number for that call. Is this possible? I am using async in the AMI command. Do I need to do something such as adding and event id to the AMI originate action then listen for response from AMI? Obviously I am a bit lost here. Thanks for any pointers toward the solution. JohnM -- _ -- 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] Fujitsu or Mitel PBX's
We are looking to find someone that is familiar with Fujitsu and Mitel PBX's. Email ru...@inline.net off list. 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 distributions
A J Stiles wrote: On Thursday 01 March 2012, Ralph Green wrote: Howdy, I have tried all of these and a few more. PBXinaFlash gave me the best results, by far. AsteriskNow produced a basic working system. I could not get any of the others configured to work at all. I should tell you my restrictions. I was evaluating these distros to see which one I could use to teach at a local computer group. I wanted to do very little configuration through the command line, And *that* is where you were going wrong. Look, the command line is a fact of life. Microsoft have spent a fortune telling you that you're not smart enough to use it. You do not have to fall for that. Are you going to sit back and let them call you stupid? Think of trying to make yourself understood in a foreign country by pointing and gesturing. There comes a point where you will actually have expended *more* effort than if you had just bitten the bullet and learned the language in the first place. since my goal was not just to get a working system, but to have something I could easily show others how to setup. Again, this is where the command line excels. Irrespective of how the user of the computer has set up the GUI -- what icon theme they have selected or how they have arranged the menus, which GUI tools are present and so forth -- the command line method will always be the same. You really aren't doing your students any favours if you are teaching them blindly to avoid what is basically the most powerful feature of a GNU/Linux system. AstLinux is another great one. It has an easy to use GUI, but allows and requires editing of the Asterisk confs to build a working system without the drawbacks of the others mentioned It also runs on small platforms, in 1 Gig ( or less ) of flash, no hard drive needed, and would allow the student to work with Asterisk without the overhead of learning Linux Building a system from scratch and source code is certainly best however John Novack -- 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
[asterisk-users] Line noise/hiss on Openvox A400P card on FXO
5. Placing ferrite cores on the phone cables. Do either of the phone lines in question have DSL on them? If so, a ferrite core (which will block common-mode RF signals) probably won't help much, if at all. DSL is a differential-mode signal, and its frequency content starts down in the tens of kHz. Ferrite cores are usually intended to block much higher frequency interference, and won't have enough inductance to help much with DSL signals. What I would suggest, is that you get yourself a couple of DSL microfilters... plug them into the A400P FXO ports, and plug the lines into the filters. These sorts of filters are designed specifically to block DSL differential- mode signals from getting into analog-phone circuits, and they will also be fairly effective against other forms of low-frequency-RF noise. -- _ -- 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] Google Voice STUN error?
I have been playing with gvoice over the past few months and it's been great except for this error that appears ONLY when my firewall is enabled: [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #0 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #1 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #2 failed error -1, retry The firewall is configured as documented here http://support.google.com/code/bin/answer.py?hl=enanswer=62464 I've also tried to find the offending packets with tcpdump but have had no luck. Anyone have any bright ideas? Thanks, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- _ -- 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] using AMI and Telnet to place calls
Since you are using AMI, I would assume you are using one of the AMI
interfaces from CPAN or somewhere. If this is the case you could do
something like this:
my $astman = new Asterisk::Manager;
$astman-user('mickey');
$astman-secret('mouse');
my $man_addr='192.168.23.172';
my $man_ok=1;
open (my $man_in, /etc/asterisk/manager.conf) or
$man_ok=undef;
if ($man_ok) {
while ($man_in) {
if ($_ =~ /^bindaddr/) {
(undef,$man_addr) = split /\=/, $_;
}
}
close $man_in;
}
$man_addr =~ s/\s//g;
( $man_addr )=( $man_addr =~ /(.*)/ );
$astman-host($man_addr);
$astman-connect || die Could not connect to .
$astman-host . !\n;
my %resp = $astman-sendcommand( Action = 'Originate',
Channel =
$extval,
Variable =
ARG1=$fileval,
Exten = $extval,
Context =
'playit',
priority = 1,
Number = 5551212
);
sleep 2;
n Do a while here to interrogate %resp
%resp = $astman-sendcommand( Action = 'Logoff');
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican
Sent: Thursday, March 01, 2012 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] using AMI and Telnet to place calls
Hello,
I am using a perl script to pull call info from a DB and place calls via
telnet and AMI, all on local machine of course. My problem is that I need
to capture any response from the carier, such as this taht appears in the
CLI:
[Mar 1 12:55:50] == Using SIP RTP CoS mark 5
[Mar 1 12:55:50] -- Got SIP response 503 No Circuit Available back
from xxx.xxx.xxx.xxx:5060
[Mar 1 12:55:50] Channel SIP/provider was never answered
and be able to relate that back to the dialed number for that call. Is this
possible?
I am using async in the AMI command. Do I need to do something such as
adding and event id to the AMI originate action then listen for response
from AMI?
Obviously I am a bit lost here.
Thanks for any pointers toward the solution.
JohnM
--
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[asterisk-users] Difference between busy / unavailable greetings in an environment with call waiting
All my phones have call waiting, so it's unlikely DIALSTATUS ever gets set to BUSY. So, I'm trying to decide what to do about the two greetings users record, busy and unavailable. If I could, I could just disable one. Then there's only one greeting, and no chance for confusion. I could probably find a way to re-record the options menu to just not tell users about it, but I'd also like to use forcegreetings=yes, which will make them record both. Any way to force them to just set one, or otherwise disable one of the greetings? Alternately, I'm open to suggestions on how to make this functionality not confusing for my users. Call waiting seems pretty ubiquitous on business phones these days, so how are people handling this issue? -- _ -- 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] Digium FXS specifications and limits Question
Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does not have this same notice. Should it? My potential application will be pushing all of 1500 feet and maybe a teeny bit more. Analog lines will be bundled into PE-89 type direct-burial rated cable in capacities ranging from 100-pair to 500-pair. All cable pairs are 24AWG. All cable is privately owned by the organization. All cable is professionally terminated on each end to grounded building entrance protection with lightning blocks / surge protection. This infrastructure has been in place and working on an existing old school digital switch with port cards that don't have lightning or surge suppression built in, just like the Digium cards. Has anyone run a similar configuration and can speak for or against such an idear? Thanks for the help and input, Jason -- _ -- 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 distributions
On 2 March 2012 01:43, A J Stiles asterisk_l...@earthshod.co.uk wrote: Look, the command line is a fact of life. Microsoft have spent a fortune telling you that you're not smart enough to use it. You do not have to fall for that. Are you going to sit back and let them call you stupid? Think of trying to make yourself understood in a foreign country by pointing and gesturing. There comes a point where you will actually have expended *more* effort than if you had just bitten the bullet and learned the language in the first place. My signature is probably applicable here (out of the fortune database)... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett -- _ -- 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 CDRs
Hi all, It disturbs me to see asterisk (v 1.6.2.10) writing CDRs even when there are 0 active channels and 0 active calls. Is there an upper limit in terms of CDRs / second that asterisk can handle? Does it queue the unwritten CDRs somewhere? Please help me clarify this confusion. 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] Line noise/hiss on Openvox A400P card on FXO
On 01/03/12 19:07, Dave Platt wrote: 5. Placing ferrite cores on the phone cables. Do either of the phone lines in question have DSL on them? If so, a ferrite core (which will block common-mode RF signals) probably won't help much, if at all. DSL is a differential-mode signal, and its frequency content starts down in the tens of kHz. Ferrite cores are usually intended to block much higher frequency interference, and won't have enough inductance to help much with DSL signals. What I would suggest, is that you get yourself a couple of DSL microfilters... plug them into the A400P FXO ports, and plug the lines into the filters. These sorts of filters are designed specifically to block DSL differential- mode signals from getting into analog-phone circuits, and they will also be fairly effective against other forms of low-frequency-RF noise. Hi Dave and thanks for the suggestion. Although the lines don't have ADSL on them (we have a separate line for that) - I've tried ADSL filters on the them anyway. I even chained two filters to see if it will have any effect. I'm afraid the noise is still the same. 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
[asterisk-users] ph_tor3_e1.c
We are running FreeTDM on a very cheap Atcom card, which used another module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta. http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence dahdi) does not unload. Sometimes the machine hangs and needs to be rebooted. This module has not been updated for the last 2 years during which the linux kernel has changed (I am told). Is there any other manufacturer of torrent card who would be using the same architecture and keeping his drivers updated ? If not, what steps do I need to take to update this driver to kernel version 2.6.32-37-server ? These are new waters and I feel so helpless :) regards, Anita regards, Anita -- _ -- 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 CDRs
Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- 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 CDRs
I've tried with batch enabled as well as disabled, it seems irrespective of the call burst I send to asterisk. CDR writes at a constant speed, not changing with the call load! On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.com wrote: Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- 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
