Re: [asterisk-users] Speed Dials Management....

2011-08-16 Thread William Stillwell
What I have done is created a special extension # (ie, 63XXX) and then created a mysql database with XXX and the number to call, then when the 63xxx extension is dialed it looks up the number in the database via agi script and completes the call. From:

[asterisk-users] use dahdi for local terminal modem access?

2011-07-22 Thread William Stillwell
I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell

Re: [asterisk-users] Goggle voice incoming dialplan

2011-06-16 Thread William Stillwell
Only GV numbers that can terminate to a Google Chat Account can be connected directly to asterisk. Otherwise you will need to get a free SIP Account, and route calls to it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of

Re: [asterisk-users] Goggle voice incoming dialplan

2011-06-16 Thread William Stillwell
Discussion Subject: Re: [asterisk-users] Goggle voice incoming dialplan Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? On Fri, Jun 17, 2011 at 11:48 AM, William Stillwell will...@stillwellsoft.com wrote: Only GV numbers

Re: [asterisk-users] Full SIP dial string

2011-06-11 Thread William Stillwell
/username:password@host/exten without success Can you help ? Thanks --- Try SIP/exten@host William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] issues.asterisk.org/jira not working

2011-06-08 Thread William Stillwell
You mean this one? https://issues.asterisk.org/jira/browse/ASTERISK-17984 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 2:17 PM To: asterisk-users Subject: [asterisk-users]

Re: [asterisk-users] asterisk 1.8 + google voice

2011-05-13 Thread William Stillwell
Im running v1.8.2.3 and not have no had this issue you speak of? I saw it once or twice, but otherwise, it works. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeremy Kister Sent: Thursday, May 12,

Re: [asterisk-users] Good by asterisk 1.4? Please not.

2011-04-19 Thread William Stillwell
Its not really had to install 1.6 or 1.8 on a test box, and see if a phone connects to it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, April 19, 2011 11:02 AM To:

Re: [asterisk-users] Google Voice receiving call problem

2011-04-16 Thread William Stillwell
You must have 1.8+ its already been posted the 1.6 didn't get a backport fix in the jabber protocol. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Saturday, April 16, 2011 3:57 AM To:

Re: [asterisk-users] Back-to-back asterisk PRI issue

2011-03-25 Thread William Stillwell
Did you check so see if the pri is up? Also, make sure wanpipe dahdi is setup correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, March 25, 2011 3:41 PM To: asterisk-users Subject:

Re: [asterisk-users] What is the most stable version of asterisk?

2011-03-23 Thread William Stillwell
I feel there all pretty stable, 1.4,1.6,1.8 depending on how deep of a feature set you want.. if you just want the bare essentials, go with 1.2, but if you want faxing/gTalk, then go with 1.8 I have been running 1.4,1.6,1.8 in production environments and have not have had any serious issues.

Re: [asterisk-users] wav files are not playing asterisk

2011-03-02 Thread William Stillwell
Your error is in front of you. format_wav.c:148 check_header: Not in mono 2 [Feb 28 22:27:07] WARNING[2736]: file.c:386 fn_wrapper: Unable to open format wav Your wav file is not in proper format. Must be mono, and at 8khz, 16bit You can resample by using this command: sox

Re: [asterisk-users] Asterisk/Skype

2011-02-25 Thread William Stillwell
Maybe something like this? [skype_chat_receieve] Exten = account,user,1,do something here? What do you see in the CLI on the incoming txt message? I just figured out how to handle a different google talk account today [google-in] Exten =

Re: [asterisk-users] Asterisk/Skype

2011-02-25 Thread William Stillwell
I am assuming that goes the same for Gtalk chat messages too? Or has nobody played with that? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson Sent: Friday, February 25, 2011 3:16 PM To: Asterisk Users Mailing

Re: [asterisk-users] Google Voice outbound Caller ID broken

2011-02-24 Thread William Stillwell
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Thursday, February 24, 2011 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Google Voice outbound Caller ID broken Anybody

Re: [asterisk-users] Paging with Polycom 3.3.x

2011-02-24 Thread William Stillwell
Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re:

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-20 Thread William Stillwell
, 2011 5:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Gtalk/Jabber Issue On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-20 Thread William Stillwell
Of William Stillwell Sent: Sunday, February 20, 2011 6:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Gtalk/Jabber Issue I was also informed it only works in 1.8, I think there was a protocol change I think that wasn't back ported to 1.6

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-20 Thread William Stillwell
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Sunday, February 20, 2011 10:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Gtalk/Jabber Issue

Re: [asterisk-users] uptime

2011-02-14 Thread William Stillwell
Sounds like a clock slip/ntp issue -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, February 14, 2011 10:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users]

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-11 Thread William Stillwell
that nothing blocks the specific traffic on your network? Any chance of taking the packet trace on your gateway? -Vladimir On 2/11/2011 1:18 AM, William Stillwell wrote: I don’t’ appear to have an jabber [] OUTGOING packets? I get just 1 incoming packet, and it just sits there, until it rings

[asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 11:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Gtalk/Jabber Issue Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
Mikhelson [mailto:v...@mikhelson.com] Sent: Friday, February 11, 2011 12:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: William Stillwell Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, Have you tried outgoing calls? What happens there? Have you restarted

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over

Re: [asterisk-users] Gtalk/Jabber Issue

2011-02-10 Thread William Stillwell
, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Call Recording audio file quality query

2011-02-08 Thread William Stillwell
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, February 08, 2011 6:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Recording audio file quality

[asterisk-users] OT: SwitchVox Mailing List?

2011-02-07 Thread William Stillwell
=4adb81c464701e0039d e21a300aa273f t=77031sid=4adb81c464701e0039de21a300aa273f William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] PRI voice optimization

2011-02-04 Thread William Stillwell
card with tata PRI lines. regards dhaval Dhavel, The TE410P doesn't have echo cancellation built in, do you have the VPMOCT128 Echo Cancellation module attached? The TE412P is the model with/Echo Cancellation hardware. William Stillwell

Re: [asterisk-users] Reducing number of Asterisk processes?

2011-01-29 Thread William Stillwell
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gilles Sent: Saturday, January 29, 2011 11:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reducing number of Asterisk processes?

[asterisk-users] Anybody ever see this before?

2011-01-27 Thread William Stillwell
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error: Should have only transmitted 0 frames! [Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error: Should have only transmitted 0 frames! I just saw it fly across my CLI. --

Re: [asterisk-users] spandsp download

2011-01-22 Thread William Stillwell
Steve, Do you have a change log for what has changed between pre17 and pre18 ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Underwood Sent: Saturday, January 22, 2011 10:49 AM To:

Re: [asterisk-users] FUNC_ODBC and ARRAY

2011-01-22 Thread William Stillwell
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Saturday, January 22, 2011 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FUNC_ODBC

Re: [asterisk-users] ReceiveFax

2011-01-20 Thread William Stillwell
This is new to me, I have a fax server using Receive Fax and gets way over 5 calls at a time. [fax-in] exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) ;exten = s,n,Set(${LOCALSTATIONID}) exten =

Re: [asterisk-users] Accessing a 'user' variable via. dialplan.

2011-01-20 Thread William Stillwell
, Setvar=VAR_1=Taco Setvar=VAR_2=Apples Setvar=VAR_3=Bannanna -- William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] AGI-Macro w/Agruments

2011-01-14 Thread William Stillwell
So, I take it nobody has a solution to this? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Friday, January 07, 2011 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk

[asterisk-users] Mail list Woes?

2011-01-09 Thread William Stillwell
Anybody notice log delays in this list, and very small amount of traffic? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] AGI-Macro w/Agruments

2011-01-08 Thread William Stillwell
OK, I need to dial a macro from AGI and needs to pass an argument. Ok, I found an bug report, but it was stated un fixable? really after 5 years? https://issues.asterisk.org/view.php?id=2470 I found this email in the archive, but no solution other then the dodgy work around?

Re: [asterisk-users] call is not going to Voicemail with 1,n

2010-12-29 Thread William Stillwell
The n = prev + 1 So you dialplan technically looks like this: exten = 1,1,Playback(transfer) exten = 1,2,Dial(${sales_support}IAX2/iaxy-322,20,jrw) exten = 1,103,Voicemail(11,b) exten = 1,104,Hangup() exten = 1,105,Voicemail(11,b) ; Right to voicemail exten = 1,106,Hangup() which in

Re: [asterisk-users] call is not going to Voicemail with 1,n

2010-12-29 Thread William Stillwell
Also, a more fancy approach [macro-dialvm] exten = s,1,NoOp(${ExTEN}|${MACRO_EXTEN}|${ARG1}) exten = s,n,Dial(SIP/${ARG1},25,t) exten = s,n,NoOp(${ARG1}) exten = s,n,NoOp(${DIALSTATUS}) exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?BUSY) exten = s,n,GotoIf($[${DIALSTATUS} = NOANSWER]?NOANSWER)

Re: [asterisk-users] Panasonic KX-TGP500 w/Asterisk

2010-12-28 Thread William Stillwell
Well, my distributor states the 550 has been discontinued, I'm going to try out the 500 and see how well it works, as it appears to be the only Sip based cordless phone that looks to be half way decent.. (not a big fan of Sip/2.4/5.8ghz Wifi Phones) -Original Message- From:

[asterisk-users] Panasonic KX-TGP500 w/Asterisk

2010-12-27 Thread William Stillwell
Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? I run a small asterisk server at home using two SPA3102s, and thinking of upgrading my cordless analog phones to something a little newer. --

Re: [asterisk-users] Recommendation for a Linux based SCADA

2010-12-20 Thread William Stillwell
Any device that you can talk to and be used in Linux can be interfaced into asterisk with the power of AGI. I have some WebRelay modules that I can remotely control via asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

Re: [asterisk-users] Specifying DID for outbound calls

2010-12-19 Thread William Stillwell
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese Sent: Sunday, December 19, 2010 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Specifying DID

Re: [asterisk-users] Specifying DID for outbound calls

2010-12-19 Thread William Stillwell
...@vitel-outbound) Of course you could add some sanity checking there to make sure that ${EXTERNAL_CALLERID} contains a value and if not default to your main DID. How would that work if a user has 3 different callerids, and the use of realtime? William Stillwell

Re: [asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?

2010-12-08 Thread William Stillwell
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Wednesday, December 08, 2010 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to quickly move on to Dahdi channels

Re: [asterisk-users] kernel: dahdi: Detected time shift.

2010-11-24 Thread William Stillwell (Lists)
. William Stillwell From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, November 24, 2010 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] kernel: dahdi: Detected time

Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem

2010-11-19 Thread William Stillwell (Lists)
Why not just use tiff2pdf ? tiff2pdf input.tif -o output.pdf William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Michael Sent: Friday, November 19, 2010 9:43 AM To: asterisk-users

Re: [asterisk-users] ISDN-FAX with Asterisk

2010-11-18 Thread William Stillwell (Lists)
. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Thorolf Godawa Sent: Thursday, November 18, 2010 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk

Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100

2010-11-03 Thread William Stillwell (Lists)
providers, you're going to get hit or miss performance with faxing. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ronny Adsetts Sent: Wednesday, November 03, 2010 8:06 AM To: Asterisk Users

[asterisk-users] doh! chan_dahdi.conf

2010-11-03 Thread William Stillwell (Lists)
immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be [Channels] GlobalOption=Value [1-24] Option=value [25-48] Option=value [49-72] Option=value William Stillwell

Re: [asterisk-users] doh! chan_dahdi.conf

2010-11-03 Thread William Stillwell (Lists)
On Wed, Nov 03, 2010 at 06:30:09PM -0400, William Stillwell (Lists) wrote: For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular channels must be placed before the channel = entry. Immediate=no Channel=1-24 Immediate=yes

Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100

2010-11-02 Thread William Stillwell (Lists)
with pure asterisk solution, we had over 100 stations, so couldn't do that quite easily. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Tuesday, November 02, 2010 12:36

[asterisk-users] Channel Bank ? Simple Switch Hangup?

2010-10-26 Thread William Stillwell (Lists)
I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get SimpleSwitch and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf:

Re: [asterisk-users] Channel Bank ? Simple Switch Hangup?

2010-10-26 Thread William Stillwell (Lists)
Nevermind, figured it out. Immediate=yes on top part of chan_dahdi.conf And in extensions.conf Exten =s,1,disa(no-password,internal) William Stillwell Systems Architect MDT Personnel, LLC. Ph. Coming soon. Fx. Coming soon. Cl. 727-638-6208 From: asterisk-users-boun

Re: [asterisk-users] Soft phones.

2010-07-22 Thread William Stillwell (Lists)
Zoiper seems to have a software update every other week, and annoys you to death on updates, and sometime the update breaks it. I am looking myself for a good windows softphone, Zoiper is nice, never tried the pay for version. -Original Message- From:

Re: [asterisk-users] MAC Address prefixes of Voip equipment

2010-07-12 Thread William Stillwell (Lists)
http://www.coffer.com/mac_find/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church Sent: Monday, July 12, 2010 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] MAC Address prefixes of Voip equipment

2010-07-12 Thread William Stillwell (Lists)
Another tool, to search by company. http://standards.ieee.org/regauth/oui/index.shtml -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C. Bailey Sent: Monday, July 12, 2010 11:53 AM To: Asterisk Users

Re: [asterisk-users] How to stop intruder from registering sip?

2010-07-01 Thread William Stillwell (Lists)
Also, technically your 101This is a salt is stronger than your SHA1 Hash. Let's say you stick with the 17 character password You are using 0-9, a-z, A-Z, and space. 0-9 = 10 a-z = 26 A-Z = 26 Space = 1 Total Possible Values = 63 17^63 = 3.2982384238829760312713680399948e+77 Your sha1 is

[asterisk-users] SIP Delay with remote stations?

2010-06-29 Thread William Stillwell (Lists)
I have several remote phones that experience a slight call delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. --

Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

2010-06-29 Thread William Stillwell (Lists)
I use SecureCRT+FX , and use ansi graphics. Putty is nice w/WinSCP as well. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 29, 2010 10:17 AM To: Asterisk Users Mailing List

Re: [asterisk-users] Dialplan for conference

2010-06-24 Thread William Stillwell (Lists)
I know on my polycom phones, I just press the conf button, dial, and then hit join, and all done, no special programming required on dialplan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread William Stillwell (Lists)
I would think AGI would be better. ? I don't think system() returns anything, except maybe a success/fail ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Monday, June 14, 2010 12:00 PM To: Asterisk

Re: [asterisk-users] Music on Hold

2010-05-26 Thread William Stillwell (Lists)
You would need to see if there is a hook flash hold. Try playing with a hook/flash ( ie do a flash, wait, then hangup phone, it may send the onhold message ) it may also ring back. Or you will have to park the call Hook flash , Dial 700 (if that's your park extension), hangup, then

Re: [asterisk-users] Softphones on thin clients...

2010-05-20 Thread William Stillwell (Lists)
Don't some thin clients run on WindowsCE or Linux/rdesktop? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, May 20, 2010 1:51 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution)

2010-05-14 Thread William Stillwell (Lists)
with spandsp/receivefax/asterisk 1.6.2.7 then asterisk 1.4.x w/pikafax. Keep up the good work :) William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Underwood Sent: Friday, May 14, 2010 12:33 PM

Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution)

2010-05-13 Thread William Stillwell (Lists)
ram, 320gb raid 0 sata Thanks to all who offered suggestions, and such, I will try this out, and hopefully should work well, as Steve Hinted to a year ago. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Need fax solution for 1.4.xx

2010-05-12 Thread William Stillwell (Lists)
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx William Stillwell (Lists) wrote: Anybody know a reliable fax solution for 1.4.30 branch? That would be HylaFAX+ along with iaxmodem http://hylafax.sourceforge.net http://iaxmodem.sourceforge.net Doug -- Ben Franklin quote: Those

Re: [asterisk-users] Need fax solution for 1.4.xx

2010-05-12 Thread William Stillwell (Lists)
something like the MyFax service. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Tuesday, May 11, 2010 2:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users

Re: [asterisk-users] Need fax solution for 1.4.xx

2010-05-12 Thread William Stillwell (Lists)
== On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists) william.stillwell-li...@ wrote: Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer

[asterisk-users] Need fax solution for 1.4.xx

2010-05-11 Thread William Stillwell (Lists)
Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there WARP appliance. NOT really looking to migrate from 1.4.x

Re: [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server

2010-05-03 Thread William Stillwell (Lists)
What ports to you have available on the ESI ? Analog Trunk Lines? Analog Station Lines? PRI? You could bridge with maybe a small 4 or 8 port FXO/FXS device depending on what you have available in on your ESI. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Converting GSM calls to SIP

2010-04-14 Thread William Stillwell (Lists)
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Wednesday, April 14, 2010 10:52 AM To: asterisk-users@lists.digium.com Subject:

[asterisk-users] Queue Member stuck in Ring+InUse?

2010-03-09 Thread William Stillwell (Lists)
Anybody work out how to fix this? Asterisk 1.4.26.3 Sip Trunk inbound - to Queuee - Outbound to two sip stations, and one sip trunk. sip trunk caller answers, queue shows ring+inuse , core show channels shows inbound/outbound after caller hanges up, no channels in use, queue still

[asterisk-users] Qeuee/Agent Question

2010-02-26 Thread William Stillwell (Lists)
What is the easiest method or can someone point me in the direction I need to look to do remote agent login.. Ie, Caller calls in with a cell or home phone, authenticates himself, select a queue to be added too, hangs up, and then any calls coming into said queue would ring their home or cell

Re: [asterisk-users] Issue with trying to dial two different servers at the same time.

2010-02-16 Thread William Stillwell (Lists)
Do a qeuee, add each as a station in the quee.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Anness Sent: Tuesday, February 16, 2010 10:04 AM To: asterisk-users@lists.digium.com Subject:

Re: [asterisk-users] IP Phone recommendation

2010-02-10 Thread William Stillwell (Lists)
Polycom 331's are also in the same price range, and offer good features as well. All my polycoms are provisions with option 66 on dhcp, and an ftp site with cfg files that are build from a mysql database from sip users table. From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime

2010-02-08 Thread William Stillwell (Lists)
Box #1 faxserver*CLI core show version Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux on 2008-08-07 20:30:54 UTC faxserver*CLI core show uptime System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds faxserver*CLI this box gets about 200 faxes a day,

Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime

2010-02-08 Thread William Stillwell (Lists)
servers, the crashing went away. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Monday, February 08, 2010 8:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Set CDR userfield for Queues

2010-02-01 Thread William Stillwell (Lists)
track your call over your asterisk system. I wrote about this in old post and submit an complete solution. Regards, On Sun, Jan 24, 2010 at 1:14 PM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: Yeah, after hours of trying Friday, I got working by a macro.. I didn't like

Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

2010-01-29 Thread William Stillwell (Lists)
) PBX (pbx2)? 3- i am using two identical dialplan's is this gonna confuse the communication process (contextes's name are duplicated over the two servers) thank you very much for making it clear for me! 2010/1/28 William Stillwell (Lists) william.stillwell-li...@ablebody.net Your inbound

Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

2010-01-28 Thread William Stillwell (Lists)
Your inbound context needs to have access to your outbound context. [iax-inbound] Include = outbound-conext [outbound-context] Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN}) Something like that. From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2

2010-01-26 Thread William Stillwell (Lists)
This is how I did it.. I have to Servers, SRV1 and SRV2 In SRV1 iax.conf [SRV1-SRV2] type=peer username=SRV1-SRV2 secret=Password1 host=IP OF SRV2 qualify=yes [SRV2-SRV1] type=user username=SRV2-SRV1 secret=Password2 context=from-iax host=IP OF SRV2 quailfy=yes If

Re: [asterisk-users] Set CDR userfield for Queues

2010-01-24 Thread William Stillwell (Lists)
23, 2010 at 12:14 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: setinterfacevar=yes Needs to be under each queue What does your dialplan end up looking like? I would like to add to mine, and stop running a cron job.. exten = 5000,1,Answer exten = 5000,n,Queue

Re: [asterisk-users] Set CDR userfield for Queues

2010-01-22 Thread William Stillwell (Lists)
Let me know if you figure it out, I am interested in this as well. Right now I have a cron job that executes this every 5 minutes.. UPDATE cdr SET userfield = MID( dstchannel, 1 , LOCATE( '-', dstchannel )-1) WHERE disposition = 'ANSWERED' AND LOCATE( '-', dstchannel ) 0 and lastapp = 'Queue'

Re: [asterisk-users] Set CDR userfield for Queues

2010-01-22 Thread William Stillwell (Lists)
setinterfacevar=yes Needs to be under each queue What does your dialplan end up looking like? I would like to add to mine, and stop running a cron job.. exten = 5000,1,Answer exten = 5000,n,Queue(5000|rn) exten = 5000,n,VoiceMail(5000,u) exten = 5000,n,Hangup -Original Message- From:

Re: [asterisk-users] Odd message: correct auth, but ...

2010-01-20 Thread William Stillwell (Lists)
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html phone is using old authentication challenge, you may have restarted asterisk, or did a sip reload, if the message is driving you batty, reboot the phone. -Original Message- From:

Re: [asterisk-users] More than a line with same extension + Polycom 320 + Provision Tool

2010-01-20 Thread William Stillwell (Lists)
I use the 331, and only have 1 line assigned, and each phone has a call limit of 10, if another call comes in, they can answer it, and it would put the other caller on hold, you can then switch between callers by using the up/down keys. -Original Message- From:

Re: [asterisk-users] help with picking out a digium card.

2010-01-20 Thread William Stillwell (Lists)
What is the configuration of the TDM400? Sangoma makes a nice card as well., I think the A200 is available in PCIe and supports from 2-4 and I think the A400 does 2-24 If you just answer 4 lines.. you could always just use a SIP Gateway, and not use any PCIe card. If you have a pbx, maybe a

[asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
I know in v1.6 its part of logger.c but I noticed this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625 However, it doesn't seem to ever been applied to any version of 1.4.x branch.. Nor can I figure out what it was applied to? This is over 3 years old, you

Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
logger.conf [general] queue_log = yes queue_log_name = queue_log Thanks, Best regards!! Cristian Arguello. - Original Message - From: William Stillwell (Lists) mailto:william.stillwell-li...@ablebody.net

Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell (Lists) Sent: Tuesday, January 19, 2010 1:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ? Yeah, I know all that.. I am just saying

[asterisk-users] Getting Answered Stations instead of Group in cdr?

2010-01-15 Thread William Stillwell (Lists)
I have a dialplan entry that takes a did, and sends it to a group of stations Dial(Sip/ExtSip/ExtSip/Ext) etc. However, cdr only shows dst = 5000 (given) and lastdata shows the dial context, however I see no cdr entry for who actually answered the phone. , I can see dstchannel as

Re: [asterisk-users] Beginners Guide to setting up a Call Centre

2010-01-15 Thread William Stillwell (Lists)
Most important thing is to PLAN your solution out.. flowcharts, understanding where calls go, etc. Project planning, and good ideas on how the calls should be handled, and coming up with testing scenarios, to make sure everything flows correctly. From:

Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4

2010-01-15 Thread William Stillwell (Lists)
Here is the 1.4.x version on centos 5 walk through. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik Sent: Friday, January 15, 2010 3:15 PM

[asterisk-users] Odd Voicemail Issue

2010-01-12 Thread William Stillwell ( Lists )
I have several extensions in the Central Timezone, the Server is in the Eastern Timezone. all the voicemail files have a datetimestamp of EST not of the tz= option under the usermail ... voicemail.conf under [general] tz=EST under [default] mailbox_a,password,,,tz=CST6CDT

Re: [asterisk-users] Odd Voicemail Issue

2010-01-12 Thread William Stillwell ( Lists )
ok, I figured it out.. tz=zonename from zonemessages all fixed. - Original Message - From: William Stillwell ( Lists ) To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 12, 2010 9:59 PM Subject: [asterisk-users] Odd Voicemail Issue I

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