Re: [asterisk-users] Auto dialer scripts and software

2013-05-23 Thread cjwstudios
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 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 pa

[asterisk-users] Auto dialer scripts and software

2013-05-17 Thread cjwstudios
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. --

Re: [asterisk-users] Confbridge examples for Asterisk 10?

2012-07-26 Thread cjwstudios
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 wrote: > cjwstudios wrote: >> >> Does anyone have any appl

[asterisk-users] Confbridge examples for Asterisk 10?

2012-07-25 Thread cjwstudios
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.ap

[asterisk-users] Audiocodes Mediant 1000 setup

2012-06-13 Thread cjwstudios
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:

Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax

2012-05-03 Thread cjwstudios
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 wrote: > Hi > > We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the > results make us sad :( > > I suppose Aste

[asterisk-users] Account code script needed.

2012-04-17 Thread cjwstudios
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://w

[asterisk-users] process_sdp: Multiple audio streams are not supported

2012-04-03 Thread cjwstudios
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

Re: [asterisk-users] New Dahdi error

2011-01-10 Thread cjwstudios
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 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. >&g

[asterisk-users] New Dahdi error

2011-01-10 Thread cjwstudios
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] E

Re: [asterisk-users] VoIP PoE phones for restaurant

2011-01-03 Thread cjwstudios
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 wrote: > Greetings, > I mailed the list regarding an intercom system som

Re: [asterisk-users] app_directory broken in 1.6

2010-01-31 Thread cjwstudios
Thanks Tilghman, that made a substantial difference. On Sun, Jan 31, 2010 at 6:18 PM, Tilghman Lesher wrote: > On Sunday 31 January 2010 18:12:15 cjwstudios wrote: > > Hello, > > > > I have separate contexts defined in voicemail.conf as follows: > > > > [ab

[asterisk-users] app_directory broken in 1.6

2010-01-31 Thread cjwstudios
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 voicema