Re: [asterisk-users] Delay in IVR
Hi, sorry for my insistence but I would your aid for my problem. Thanks. -- Salvatore. - Original Message - From: Sasa s...@shoponweb.it To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 03, 2010 9:51 AM Subject: Re: [asterisk-users] Delay in IVR Hi, in trixbox I don't know what create an extension with letter but only with number. Thanks. -- Salvatore. - Original Message - From: Kingsley Tart kings...@skymarket.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 02, 2010 5:34 PM Subject: Re: [asterisk-users] Delay in IVR On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote: On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote: HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. I know nothing of Trixbox but I had a problem with my own dialplan where there was a delay with the user selecting 0 from my IVR menu. It turned out that because my extensions all started with 0 (they were real phone numbers), asterisk thought that the caller might be starting to type one of the valid extensions and so waited for the timeout (digit timeout I think) before it went further. To see if that's your problem, try seeing whether a menu option that won't match the start of any of your defined extensions happens more quickly. I got around it by having the IVR in a different context where the extensions started with a letter, so no entered digits would match. Salvatore, Did this help? -- Cheers, Kingsley. -- _ -- 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
[asterisk-users] Out of Office
I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. -- _ -- 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] AMI Queue information about incoming call's channel before link
Hi, I'm developing an application using AMI and I need to get information about incoming call _before_ queue member answers it. I'm using static members (queue is pretty simple) and don't want to use chan_agent (I think AgentCalled event will do what I'm looking for). Here is what I'm getting now: 1. Newchannel event for incoming call followed by Newstate and Join. All these events can be identified with uniqueid 001 2. Newchannel event for outgoing channel to queue member followed by Newstate State: Ringing = that's what I need. But this channel (of course) has another uniquied 002. 3. Link. That's the place where channels are get bridged and I can find out which member (by channel) answered the call (002 answered 001 in this example). The problem is that I need to get information about call 001 at stage 2 that is currently possible only at stage 3, when channels get bridged. Any ideas? -- Alexandr Krylovskiy -- _ -- 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] Out of Office
How annoying this is. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-09 3:59 AM, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. -- _ -- 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] get Asterisk version from within dialplan
Simple enough: How can I get Asterisk version from within my dialplan? (preferably without calling an AGI script that parses asterisk -rx show version) Is it available as a global variable? Vieri -- _ -- 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] PSTN-IVR call
hi all, I am calling a PSTN and trying to transfer it to another asterisk server through exec_dial function. $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL); Though this is the function written by me in a file inbound.php which is called when an extension is dialled. When ever i dial through PSTN it gives beeps sound, but without this line program runs smoothly. Can someone help??? -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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] Out of Office
Let's all send John and Mary an email to tell them how thoughtful Doug is and you can bet he will either turn off or modify his rule :-) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, June 09, 2010 4:37 AM To: d...@accessgate.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out of Office How annoying this is. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-09 3:59 AM, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. -- _ -- 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] PSTN-IVR call
Since you have some inherent timeouts due to technology jumps, you might try $agi-exec_dial(SIP,ww2001:j0...@172.26.48.62:5060,NULL,NULL,NULL); To make asterisk wait 1 second before trying the call. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nikhil singhania Sent: Wednesday, June 09, 2010 7:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PSTN-IVR call hi all, I am calling a PSTN and trying to transfer it to another asterisk server through exec_dial function. $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL); Though this is the function written by me in a file inbound.php which is called when an extension is dialled. When ever i dial through PSTN it gives beeps sound, but without this line program runs smoothly. Can someone help??? -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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] OT - Astmanproxy download broken ?
Hi, Is Astmanproxy still downloadable ? At the moment, I can't download anything. I'm usually using this http://github.com/davetroy/astmanproxy/tarball/masterURL I can use a previous tar file but I would be pleased to know if I should do something around this issue or not. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get Asterisk version from within dialplan
Starting with version 1.6.x there is a VERSION function that I think will give you the version number. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 9, 2010, at 5:19 AM, Vieri wrote: Simple enough: How can I get Asterisk version from within my dialplan? (preferably without calling an AGI script that parses asterisk -rx show version) Is it available as a global variable? Vieri -- _ -- 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] Delay in IVR
I use to use trixbox its basically asterisk with free pbx. What are your extension numbers? Ring group number? What processor are you using? The more info the better. When I used trixbox I never had this problem. It could be DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind of phones. What is the drtmf setting on your phones? What kind of phone are you testing this with? I always have one test sip trunk that I know works great for testing. Its not likely in the code of the ivr. The problem is in how you setup everything else that leads to trunk. Respectfully Michael D Mosier Ftoc Certified On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote: Hi, sorry for my insistence but I would your aid for my problem. Thanks. -- Salvatore. - Original Message - From: Sasa s...@shoponweb.it To: Asterisk Users Mailing List - ... Sent: Thursday, June 03, 2010 9:51 AM Subject: Re: [asterisk-users] Delay in IVR Hi, in trixbox ... -- _ -- 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] SIP Witch
Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch GNU SIP Witch - GNU Telephony http://www.gnutelephony.org/index.php/GNU_SIP_Witch Features/SIP Witch Domain Telephony http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony Secure VoIP, GNU SIP Witch, and replacing Skype with free software http://www.linuxtoday.com/it_management/2009082702235OSNT Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Can one adjust the voicemail-menu when using VoiceMailMain() ?
I have commented out case 5, case 2 and case 3, leaving case 1, 4,6,7,8,9. But when I press 1 on the menu, I hear: I'm sorry, I did not understand your response if (play_auto) { cmd = '1'; } else { cmd = vm_intro(chan, vmu, vms); } vms.repeats = 0; vms.starting = 1; while ((cmd -1) (cmd != 't') (cmd != '#')) { /* Run main menu */ switch (cmd) { case '1': vms.curmsg = 0; /* Fall through */ /* commented out from here case '5': cmd = vm_browse_messages(chan, vms, vmu); break; snip code if (vms.repeats 3) cmd = 't'; } } if (cmd == 't') { cmd = 0; vms.repeats = 0; } break; commented out till here */ Jonas. On 06/07/2010 04:22 PM, Glenn O Larsen wrote: On Mon, Jun 7, 2010 at 2:15 PM, Jonas Kellensjonas.kell...@telenet.be wrote: I made some changes to app_voicemail.c and recompiled asterisk. Now my caller is only presented with the menu-choice I want. However, the caller can still give another dtmf-input and be taken to that specific menu. How can I disable dtmf-input 2,3,4 if I only want the menu behind option 1 available ? In the C code, find: case '2': /* Change folders */ if (useadsi) adsi_folders(chan, 0, Change to folder...); cmd = get_folder2(chan, vm-changeto, 0); then add the lines: + cmd = 0; /* Go back to root menu */ + break; /* don't continue */ so it look like this: case '2': /* Change folders */ cmd = 0; /* Go back to root menu */ break; /* don't continue */ if (useadsi) adsi_folders(chan, 0, Change to folder...); cmd = get_folder2(chan, vm-changeto, 0); add the same lines to case 3 and 4.. Did that help? -- _ -- 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] Delay in IVR
Hi, here information request: extension number is 100/101 ring group number is 600 cpu : Intel(R) Pentium(R) D CPU 3.00GHz 3 GHz On another voip machine (always with Trixbox) I haven't this problem, I have tried with another phones and with XLite I have always this problem. About DTMF in SIP trunk I have (in USER Details) this parameter: dtmfmode=rfc2833 Another sip trunk configuration is: PEER Details: secret=yqyqyq nat=no context=from-pstn host=x.x.x.x insecure=very type=friend username=yyy In USER Details: canreinvite=no context=from-trunk dtmfmode=rfc2833 insecure=very nat=no port=5060 type=user I hope that informtions are enough for resolve my problem Thanks. -- Salvatore. - Original Message - From: mike mosier trixbo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 09, 2010 3:36 PM Subject: Re: [asterisk-users] Delay in IVR I use to use trixbox its basically asterisk with free pbx. What are your extension numbers? Ring group number? What processor are you using? The more info the better. When I used trixbox I never had this problem. It could be DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind of phones. What is the drtmf setting on your phones? What kind of phone are you testing this with? I always have one test sip trunk that I know works great for testing. Its not likely in the code of the ivr. The problem is in how you setup everything else that leads to trunk. Respectfully Michael D Mosier Ftoc Certified On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote: Hi, sorry for my insistence but I would your aid for my problem. Thanks. -- Salvatore. - Original Message - From: Sasa s...@shoponweb.it To: Asterisk Users Mailing List - ... Sent: Thursday, June 03, 2010 9:51 AM Subject: Re: [asterisk-users] Delay in IVR Hi, in trixbox ... -- _ -- 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] PSTN-IVR call
On Wed, 9 Jun 2010, nikhil singhania wrote: I am calling a PSTN and trying to transfer it to another asterisk server through exec_dial function. $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL); Though this is the function written by me in a file inbound.php which is called when an extension is dialled. If you enable AGI debugging, this may give you a clue. Posting the console output may help someone help you. -- 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
Re: [asterisk-users] SIP Witch
I checked out the sites and can't figure out what this thing is! (Without delving into the documentation). From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew J. Roth [mr...@imminc.com] Sent: Wednesday, June 09, 2010 9:49 AM To: Asterisk Users List Subject: [asterisk-users] SIP Witch Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch GNU SIP Witch - GNU Telephony http://www.gnutelephony.org/index.php/GNU_SIP_Witch Features/SIP Witch Domain Telephony http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony Secure VoIP, GNU SIP Witch, and replacing Skype with free software http://www.linuxtoday.com/it_management/2009082702235OSNT Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Limit total length of calls to a specifig SIP peer
On 08/06/2010 19:19, Steve Edwards wrote: The ONLY way (how's that for humble) to do this in a reliable and robust method is to use a real database. Personally, I like MySQL and I prefer to do database work in an AGI in a compiled language like C. Maintaining the accumulated duration in a global variable will fail if you need to restart Asterisk at any time. A global variable will also fail if you have more than 1 call finish at the same time. Parsing log files is guaranteed to be a resource pig and still has race conditions. Hi, I'm gonna follow your advice and store the CDR in a PostgreSQL database. It will allow to easily plug an AGI script to it. Thanks Laurent -- _ -- 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] Out of Office
Good idea. I am emailing them right now. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-09 9:11 AM, Danny Nicholas da...@debsinc.com wrote: Let’s all send John and Mary an email to tell them how thoughtful Doug is and you can bet he will either turn off or modify his rule J -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Wednesday, June 09, 2010 4:37 AM *To:* d...@accessgate.net; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Out of Office How annoying this is. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-09 3:59 AM, do... -- _ -- 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] 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse
Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The call-limit option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. The limitonpeer configuration option is now renamed to counteronpeer. As i've experienced some problem with 1.4 release about call-limit, i'd like to test this new counteronpeer functionality, but how to handle the ringinuse parmeter in queues.conf ? Basically i need that a sip user can make and receive more than one call (like a call-limit 3 setting) but i don't want that this interface rings if it is in a queue. Is it possible to do that? How? Thanks to all -- /*/ nik600 http://www.kumbe.it -- _ -- 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] own Caller ID
Just is PRI line you can do it.. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Tue, 8 Jun 2010 12:44:07 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] own Caller ID On Tue, 8 Jun 2010, taimur hasan wrote: I want to use my own caller id, instead of the caller id of PSTN line, for the outbound calls through DAHDI channel. Is there any way ?? It depends on your technology (POTS, PRI, etc) and your provider. Tell your provider you want to set the outgoing caller ID and see what their response is. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [compat] section in asterisk.conf : compatibility with pipe delimiter
Dear all after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt file) the | delimiter is not working by default. I've added a compat section in asterisk.conf a [options] dontwarn = yes [compat] pbx_realtime=1.4 res_agi=1.4 app_set=1.4 And restarted Asterisk, but i still have problem to have the | delimiter working, [Jun 9 23:20:54] DEBUG[11744]: pbx.c:3122 pbx_extension_helper: Launching 'Queue' -- Executing [...@queues:4] Queue(SIP/PL1999-0003, queue_130) in new stack [Jun 9 23:20:54] DEBUG[11744]: app_queue.c:4804 queue_exec: NO QUEUE_PRIO variable found. Using default. [Jun 9 23:20:54] DEBUG[11744]: app_queue.c:4841 queue_exec: queue: queue_130, options: (null), url: (null), announce: (null), expires: 0, priority: 0 [Jun 9 23:20:54] WARNING[11744]: app_queue.c:4853 queue_exec: Unable to join queue 'queue_130' It seems that Asterisk ignores the | delimiter, if i try with the comma it works. Reading the the upgrade file it seems that the pbx_realtime should affect also the extension.conf settings... where am i wrong? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it -- _ -- 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 Witch
Matthew J. Roth wrote: Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch GNU SIP Witch - GNU Telephony http://www.gnutelephony.org/index.php/GNU_SIP_Witch Features/SIP Witch Domain Telephony http://fedoraproject.org/wiki/Features/SIP_Witch_Domain_Telephony Secure VoIP, GNU SIP Witch, and replacing Skype with free software http://www.linuxtoday.com/it_management/2009082702235OSNT Michelle Dupuis wrote: I checked out the sites and can't figure out what this thing is! (Without delving into the documentation). Michelle, It was a bit unclear to me, as well. That's why I was curious if anyone was actually using it. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Can one adjust the voicemail-menu when using VoiceMailMain() ?
At 4:04 PM on 09 Jun 2010, Jonas Kellens wrote: I have commented out case 5, case 2 and case 3, leaving case 1, 4,6,7,8,9. But when I press 1 on the menu, I hear: I'm sorry, I did not understand your response Looks like someone broke the first rule of Optimization Club[1]. I think you need to copy these two lines from case 5 into case 1: cmd = vm_browse_messages(chan, vms, vmu); break; It just so happens that cases 1 and 5 run the same command, so whoever wrote it took advantage of that, optimizing the size of the binary while reducing maintainability. [1] http://perlbuzz.com/mechanix/2008/02/the-rules-of-optimization-club.html if (play_auto) { cmd = '1'; } else { cmd = vm_intro(chan, vmu, vms); } vms.repeats = 0; vms.starting = 1; while ((cmd -1) (cmd != 't') (cmd != '#')) { /* Run main menu */ switch (cmd) { case '1': vms.curmsg = 0; /* Fall through */ /* commented out from here case '5': cmd = vm_browse_messages(chan, vms, vmu); break; snip code if (vms.repeats 3) cmd = 't'; } } if (cmd == 't') { cmd = 0; vms.repeats = 0; } break; commented out till here */ -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- 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] Out of Office
I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. -- _ -- 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] CDR in case of CallForwarding
Hello users, i am looking for a solution in terms of CDR for the outbound only call. presently i have the following setup. //extensions.conf [from-outside] exten = _X.,1,NoOp(IncomingCall) exten = _X.,n,BackGround(choce.wav) exten = _X.,n,WaitExten(5) exten = _X.,n,Hangup exten = _1XX.,n,NoOp(1XX series Dialing) exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg) exten = _1XX.,n,NoOp(${DIALSTATUS}) exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end) exten = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER FOR THE us...@ougoingprovider,60,r) exten = _1XX.,n(end),Hangup() exten = _2XX.,n,NoOp(2XX series Dialing) exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg) exten = _2XX.,n,NoOp(${DIALSTATUS}) exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end) exten = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER FOR THE us...@ougoingprovider,60,r) exten = _2XX.,n(end),Hangup() //sip.conf [outgoingprovider] username=X secret=y port= host=dfdfddf fromuser= - i am planning to take the number of calls made and the minutes spent incase of mobile call forwarding as it uses my outbound trunk by giving the accountcode set to a particular call. - but i am getting the total call (sip call + mobile call) as a single record in my cdr record for a given accountcode. - i need to get something like SIP/mobilenumber either in lastdata or dstchannel associated accountcode as a separate cdr entry. i tried with disabling cdr using NoCDR for the SIP call but for the mobile call if i use ResetCDR() also i am totally losing the callrecord. - i tried with the ForkCDR() too but of no use.. is my requirement can be fulfilled by tweaking some changes in the extensions.conf functions/applications?? please advise as i need to bill the user for the outbound calls only... any help is sincerely appreciated. thanks in advance. srinivas -- _ -- 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] CDR in case of CallForwarding
Hello I have also became like this problems and have found solution to make outgoing calls via local channel, and now if my customer do a transfer, I can calculate extra international outgoing calls. -- 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 srinivas Antarvedi wrote: Hello users, i am looking for a solution in terms of CDR for the outbound only call. presently i have the following setup. //extensions.conf [from-outside] exten = _X.,1,NoOp(IncomingCall) exten = _X.,n,BackGround(choce.wav) exten = _X.,n,WaitExten(5) exten = _X.,n,Hangup exten = _1XX.,n,NoOp(1XX series Dialing) exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg) exten = _1XX.,n,NoOp(${DIALSTATUS}) exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end) exten = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER FOR THE us...@ougoingprovider,60,r) exten = _1XX.,n(end),Hangup() exten = _2XX.,n,NoOp(2XX series Dialing) exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg) exten = _2XX.,n,NoOp(${DIALSTATUS}) exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION | ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end) exten = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER FOR THE us...@ougoingprovider,60,r) exten = _2XX.,n(end),Hangup() //sip.conf [outgoingprovider] username=X secret=y port= host=dfdfddf fromuser= - i am planning to take the number of calls made and the minutes spent incase of mobile call forwarding as it uses my outbound trunk by giving the accountcode set to a particular call. - but i am getting the total call (sip call + mobile call) as a single record in my cdr record for a given accountcode. - i need to get something like SIP/mobilenumber either in lastdata or dstchannel associated accountcode as a separate cdr entry. i tried with disabling cdr using NoCDR for the SIP call but for the mobile call if i use ResetCDR() also i am totally losing the callrecord. - i tried with the ForkCDR() too but of no use.. is my requirement can be fulfilled by tweaking some changes in the extensions.conf functions/applications?? please advise as i need to bill the user for the outbound calls only... any help is sincerely appreciated. thanks in advance. srinivas -- _ -- 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