Re: [asterisk-users] Auto dialer scripts and software
As long as you're dialing a screened registered voter list and don't call .gov or .edu, you're fine. On Wed, May 22, 2013 at 5:54 AM, Don Kelly d...@donkelly.biz wrote: Calls on behalf of political candidates are generally legal--even to people on the do not call lists. It doesn't seem to be possible to pass legislation preventing them. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Wednesday, May 22, 2013 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Auto dialer scripts and software On 22/5/13 10:54 am, A J Stiles wrote: You do know that sort of thing is against the law -- or at least requires a permit from the authorities -- in most civilised countries, right? And it's worth adding that even if it is legal in your country, you're almost guaranteed to offend/annoy your target audience. Recorded calls always do. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Auto dialer scripts and software
A friend asked me for help to auto-dial and play a prerecorded message for a political campaign. I've briefly googled auto dialer scripts but haven't seen one that really stands out. Are there any free or cheap auto dial solutions that you nice folks recommend? Thanks in advance. -- _ -- 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] Confbridge examples for Asterisk 10?
Hi Doug, I did find the following on voip-info. http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge It's somewhat rudimentary but it does work. Thanks, C On Thu, Jul 26, 2012 at 2:36 AM, Doug Lytle supp...@drdos.info wrote: cjwstudios wrote: Does anyone have any application examples for Confbridge in Asterisk 10? I'm looking for such examples as well. But just a note, meetme is still available if you're compiling from source, you just have to enable it. 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
[asterisk-users] Confbridge examples for Asterisk 10?
Does anyone have any application examples for Confbridge in Asterisk 10? I'm just looking for simple ad-hoc functionality similar to meetme in 1.8. Thank you in advance. -- _ -- 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] Audiocodes Mediant 1000 setup
Looking for help with an initial config of a Mediant 1000 with single T1/PRI. Need to route calls to an Asterisk server as well as a fax server. Please email me offlist. 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 Vs FreeSWITCH for Fax
If you're going full time hosted fax you will ultimately end up buying a t.38/sip gateway like an Audiocodes Mediant. On Thu, May 3, 2012 at 5:27 AM, Anita Hall anita.h...@simmortel.com wrote: Hi We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the results make us sad :( I suppose Asterisk also has the option of using spandsp or a commercial version from Commetrex. What are your experiences with receiving Fax on spandsp or commetrex on Asterisk ? Does it really matter whether I use Asterisk or FreeSWITCH ? 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 -- _ -- 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] Account code script needed.
Looking for quotes on a very simple script that will require a pin number before allowing a call to be placed. The pin number would be recorded to their mysql CDR. 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
[asterisk-users] process_sdp: Multiple audio streams are not supported
Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not supported in 1.8, but is there possibly a way for multiple audio streams to be supported? Thank you very much. -- _ -- 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 Dahdi error
I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4, using the wct4xxp module. All operations appear normal however I noticed an error repeating occasionally on the console. [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ROSE RETURN ERROR: [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 INVOKE ID: 11 [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ERROR: General: Not Subscribed A google search produced no result. Any ideas would be appreciated, 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
Re: [asterisk-users] New Dahdi error
Shaun, I'm using libpri-1.4.11.5. Thank you for the response. On Mon, Jan 10, 2011 at 2:05 PM, Shaun Ruffell sruff...@digium.com wrote: On 1/10/11 3:03 PM, cjwstudios wrote: I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4, using the wct4xxp module. All operations appear normal however I noticed an error repeating occasionally on the console. [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ROSE RETURN ERROR: [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 INVOKE ID: 11 [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ERROR: General: Not Subscribed A google search produced no result. Any ideas would be appreciated, thank you. What version of libpri are you using? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] VoIP PoE phones for restaurant
Andy, The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the 2626 can support vlans you can isolate data and voice. Make sure to spec a UPS on the PoE switch. On Mon, Jan 3, 2011 at 8:30 AM, Andy Graybeal andy.grayb...@casanueva.comwrote: Greetings, I mailed the list regarding an intercom system some months ago and we came to the conclusion that I should purchase a Polycom 501 phone. I'm now considering the purchase for this year, and I'm now wondering between the Polycom 501 and the 320 for the intercom. I don't need the spare ethernet on the phone because I would like to have my voice network separate from my regular LAN. Which one would be easier to use, the 501 or the 320? I want PoE, were these both made before PoE was standardized and do I need a special cable? Can I make this cable myself? In the future we plan to have 7 phones in the house. I'm considering what kind of PoE switch I should purchase. I have 3 PoE access points (for two separate LANs). I've been considering th HP ProCurve 2610-24/12PWR Switch (J9086A) ( http://h10010.www1.hp.com/wwpc/il/en/sm/WF06b/12883-12883-3445275-427605-427605-3751584-3658873.html) It's got 12 PoE ports, it's managed, and it looks like I can pick one up for under $500. Any help is appreciated. -Andy -- _ -- 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] app_directory broken in 1.6
Hello, I have separate contexts defined in voicemail.conf as follows: [abcdental] 100 = 1234,John Doe And call application directory using the following syntax: exten = 1,1,Directory(abcdental[,abcdental,f]) However since I migrated from 1.4 to 1.6, app_directory no longer parses the voicemail.conf to match the full name of the mailbox. App_directory only matches directory names based on the entries registered to the default context (voicemail show users) Therefore my workaround has been to define the users in users.conf; however the problem with that is that the fullnames in users.conf only register to the default, which means that users from multiple contexts are returned when only the specified context should be returned. It seems that the first argument of the app directory vm-context Directory([vm-context][,dial-context[,options]]) is broken as app_directory will only return matches from the default context regardless of vm-context specified. Any thoughts are appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_directory broken in 1.6
Thanks Tilghman, that made a substantial difference. On Sun, Jan 31, 2010 at 6:18 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 31 January 2010 18:12:15 cjwstudios wrote: Hello, I have separate contexts defined in voicemail.conf as follows: [abcdental] 100 = 1234,John Doe And call application directory using the following syntax: exten = 1,1,Directory(abcdental[,abcdental,f]) Uh, the square brackets in the help message means that that part is optional. Including the square bracket literally is probably why you're getting bad results. Try just: Directory(abcdental,abcdental,f) -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
