[asterisk-users] problem with voicemail contexts

2010-07-05 Thread Cassius Smith
for the mailbox. I am flummoxed. Any ideas welcome! Cassius Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] problem with voicemail contexts

2010-07-05 Thread Cassius Smith
OK, feeling very stupid right now. The test mailbox had delete=yes option set. All cleared up; sorry for cluttering up the list. Cassius snip Now, however, I don't get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says

[asterisk-users] Does SIP limit to 3-way conference?

2010-07-22 Thread Cassius Smith
other participant endpoint into the conference, but no more. I know I can (and will) use MeetMe to do large conferences. My question is - am I forced to do so by SIP? Or am I missing something? Thanks! Cassius Smith

[asterisk-users] Peculiar Polycom IP6000 behavior

2010-07-27 Thread Cassius Smith
The above entry works, but: [SPIDR-3749](caryspider) mailbox=3...@default This one doesn't. [caryspider] looks like this: [caryspider](!) type=friend context=users host=dynamic secret=xx Any ideas? I'm stumped. Cassius Smith

[asterisk-users] Caller ID issue

2010-08-02 Thread Cassius Smith
Hi list, I'm having a problem with CallerID names not showing up when calls come in. I have dialplan code to store the callerid(name) away and it is blank (null). However, the voicemail variable ${VM_CALLERID} has the name field populated. For example, here is some of the dialplan code: 2.

Re: [asterisk-users] Caller ID issue

2010-08-02 Thread Cassius Smith
Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. Ciao, Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. --

[asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932

2010-08-14 Thread Cassius Smith
Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the

Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932

2010-08-14 Thread Cassius Smith
, are the extensions set to be in the same pickupgroup? regards ron On 8/15/10 7:01 AM, Cassius Smith wrote: Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying

Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932

2010-08-16 Thread Cassius Smith
). Cassius -Original Message- From: Cassius Smith cass...@cassius.org Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users

Re: [asterisk-users] Caller ID issue

2010-08-19 Thread Cassius Smith
Sorry for the delay - I lost this message in the middle of a digest. I tried Answer(2000) and was getting an annoying warning: [Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to DAHDI/1-1 So I changed it back to Wait(2). I'll try

Re: [asterisk-users] Dahdi install gone wrong

2010-08-23 Thread Cassius Smith
* -Original Message- * From: Todd Reese trees...@gmail.com * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com * To: asterisk-users@lists.digium.com * Subject: [asterisk-users] Dahdi install gone wrong

Re: [asterisk-users] High volume BLF - Suggestions?

2010-09-13 Thread Cassius Smith
Steve I have 64 channels being monitored with an SPA962 with two SPA932 sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very happy with this. Latest firmware is a must. HTH Cassius Smith -- _ -- Bandwidth

Re: [asterisk-users] Help me Out!!!!

2010-09-15 Thread Cassius Smith
Clearly, if Word cannot explain the anguish in his heart, Mr. Fugina should be using OpenOffice! Cheers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] 3rd party app store

2010-09-21 Thread Cassius Smith
to advertise their free (!) entry points for Switchvox and FFA. Asterisk training support - I have no problem with those either. The support and training are pay-for products, but are a big help to the community also. My $0.02. Cassius Smith

[asterisk-users] advice re: Page() application

2010-10-13 Thread Cassius Smith
needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith

Re: [asterisk-users] advice re: Page() application

2010-10-13 Thread Cassius Smith
ith no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

[asterisk-users] IAX2 works one direction, but not the other...

2010-10-17 Thread Cassius Smith
I'm having trouble getting an IAX2 connection between a couple of servers. Ican make calls from server B to server A, but when I call from Server A to serverB, I get "No authority found".On ServerA I am running Asterisk 1.6.2.9On ServerB I'm running 1.6.2.13Any hints for me? The registrations in

Re: [asterisk-users] IAX2 works one direction, but not the other...

2010-10-18 Thread Cassius Smith
BTW I apologize for the double send. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] what interface for ISDN-10/20/30?

2010-10-27 Thread Cassius Smith
Hello all,I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of

[asterisk-users] Door Contacts via Asterisk?

2010-11-15 Thread Cassius Smith
. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Door Contacts via Asterisk?

2010-11-15 Thread Cassius Smith
Thanks to all for these replies. I appreciate the variety and this is a great example of the community supporting one another. I sent this in last night and awoke to a broad set of replies! Thanks all - I will post again once I decide on a solution. Cassius Smith On 11/15/10 9:09 PM, Sherwood

Re: [asterisk-users] asterisk and cisco 7970 - multiple lines

2010-11-22 Thread Cassius Smith
Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date:

Re: [asterisk-users] asterisk and cisco 7970 - multiple lines

2010-11-22 Thread Cassius Smith
I have done something similar; I am using SIP load 8.5.2. I use port 5060 on both line buttons. Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com Date: Mon, 22 Nov 2010 13:24:41 -0600 To: Cassius Smith cass...@cassius.org Cc

[asterisk-users] SPA942 on speaker phone does not hang up?

2010-11-24 Thread Cassius Smith
Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence

Re: [asterisk-users] SPA942 on speaker phone does not hang up?

2010-11-25 Thread Cassius Smith
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, November 24, 2010 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA942 on speaker phone does not hang up? Hello all, I am using Linksys SPA942 in my current installation

Re: [asterisk-users] SPA942 on speaker phone does not hang up?

2010-11-25 Thread Cassius Smith
Premature reply. It did fix the first issue. Now when I ring that phone I get busy here from the phone, and the call goes straight to voicemail per dialplan. Maybe another parameter in addition to Reorder Delay? From: Cassius Smith cass...@cassius.org Date: Thu, 25 Nov 2010 10:34:25 +0100

Re: [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk

2011-01-02 Thread Cassius Smith
CallFwd should be one of the soft keys on your Cisco phones. Are you re-flashing the Cisco phones with SIP? -Cassius On 1/2/11 3:50 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to configure the buttons in the Cisco IP Phones to be used for different functionalities like Call

[asterisk-users] TDM410 and DSL

2011-01-06 Thread Cassius Smith
Hi all, I have a system installation in Guam with two trunks. One has a DSL service riding on it with the usual filter. That channel however keeps throwing alarms. I bypassed the filter and it stopped throwing alarms, but of course the high frequencies annoy the users. I swapped the filters and

[asterisk-users] Question about EuroBRI final 2 digits

2011-02-03 Thread Cassius Smith
calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Question about EuroBRI final 2 digits

2011-02-13 Thread Cassius Smith
And when the Austria team gets to the office in the morning they will test it. (BTW changed TIMEOUT(digits) to TIMEOUT(digit)). Cassius On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom

[asterisk-users] no progress indication

2011-02-18 Thread Cassius Smith
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as

Re: [asterisk-users] no progress indication

2011-02-20 Thread Cassius Smith
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote: On 11-02-18 03:59 PM, Cassius Smith wrote: I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I

Re: [asterisk-users] Need to buy the Digium card, to confirm

2011-02-27 Thread Cassius Smith
The X1 card should seat in the X4 or X8 slots. Check out: http://computer.howstuffworks.com/pci-express1.htm HTH Cassius Smith On 2/26/11 4:33 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; My server and its slots written in it the following so I need to know which card to order it (I

Re: [asterisk-users] Play different voice-mail messages based on certain conditions

2011-03-22 Thread Cassius Smith
,Goto(2,1) exten = s,n(wrong),Playback(something-terribly-wrong) exten = s,n,Playback(goodbye) exten = s,n,Hangup() Hopefully this is enough to get you started. Cassius Smith -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering

2011-05-06 Thread Cassius Smith
Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Cassius Smith
On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be

Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering

2011-05-10 Thread Cassius Smith
fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith

[asterisk-users] lead time for RPM's?

2011-05-12 Thread Cassius Smith
Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm

Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering

2011-05-19 Thread Cassius Smith
2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Page() bumps user out of a call

2011-06-14 Thread Cassius Smith
Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The

Re: [asterisk-users] Page() bumps user out of a call

2011-06-14 Thread Cassius Smith
On 6/14/11 4:37 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On 6/14/11 4:25 PM, Russ Meyerriecks wrote: On 6/14/11 9:26 AM, Cassius Smith wrote: Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use

Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files

2011-06-16 Thread Cassius Smith
Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith -- On 6/16/11 4:59 AM, bilal ghayyad bilmar

[asterisk-users] single keypress short-circuits to invalid extension handler

2011-07-06 Thread Cassius Smith
, so I'm pretty flummoxed by thisŠ Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit s/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1

Re: [asterisk-users] single keypress short-circuits to invalid extension handler

2011-07-06 Thread Cassius Smith
On 7/6/11 3:20 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 4:37 AM To: Asterisk Users Mailing List - Non

Re: [asterisk-users] References customers

2011-07-10 Thread Cassius Smith
What do you mean by customers? Are you looking for testimonials from satisfied users? -- On 7/10/11 11:53 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How can I find a references customers that used Asterisk as IP Telephony or Call Center or IVR? In which link they are mentioned?

[asterisk-users] Receptionist Extension cannot be Pickup()'ed

2011-08-05 Thread Cassius Smith
Hello all, I am struggling with an annoying problem. I have an installation with a small number of Grandstream GXP2010 endpoints. Each endpoint has all the extensions programmed into the phone for BLF - for instant pickup, transfer or speed dial. Except for the Receptionist phone, which is

Re: [asterisk-users] Receptionist Extension cannot be Pickup()'ed

2011-08-05 Thread Cassius Smith
top posting on purpose I neglected to say ­ all the extensions can be picked up remotely by the other endpoints, EXCEPT the receptionist phone x3100. When calls go to that station, they cannot be picked up. Sorry for the necessity to post twice. /top posting on purpose From: Cassius Smith cass

Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

2011-08-16 Thread Cassius Smith
Agree -- make sure you are at the latest firmware. ALSO: If you have provisioning enabled, and have a duplicate line in your xml files, that will cause a reboot. Cheers, Cassius Smith On 8/15/11 1:46 PM, C F shma...@gmail.com wrote: I have 3 Linksys/Cisco 504G phones they keep restarting

Re: [asterisk-users] Dialout from MeetMe to another conference (Asterisk 1.4)

2011-10-10 Thread Cassius Smith
On 10/10/11 10:40 AM, Josh Freeman cpe.jfree...@gmail.com wrote: Hello, I'm looking at a scenario in which, to make it work, I'd need to dial into a remote conference from within a local MeetMe room. That might include being able to dial a conference code after the call to the remote system

[asterisk-users] Asterisk12Beta- configure script/uuid missing??

2013-10-18 Thread Cassius Smith
(this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting same error. Anyone else run into this? How did you get around it? cheers, Cassius Smith

[asterisk-users] SOLVED: Asterisk12Beta- configure script/uuid missing??

2013-10-19 Thread Cassius Smith
On Fri, Oct 18, 2013 at 03:16:08PM -0400, Cassius Smith wrote: Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: … checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking

[asterisk-users] Ast12 issue missing library file??

2013-10-23 Thread Cassius Smith
Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root@Asterisk12 ~]# asterisk -rvvv asterisk: error while

[asterisk-users] missing asterisk now rpm for centos5

2014-08-14 Thread Cassius Smith
thanks Cassius Smith -- _ -- 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