Re: [Asterisk-Users] *@Home .6 adding a outside number to a group

2005-03-21 Thread Mike
This sounds like an asterisk @ home issue. Not an asterisk issue. Asterisk at home uses a GUI that limits what asterisk can do, look at the config files it creates in (/etc/asterisk) and voip-info.org Michael On Mon, 21 Mar 2005, David Shaw wrote: Hello, I tried to add an outside number (my

Re: [Asterisk-Users] Asterisk compare with Skype

2005-03-25 Thread Mike
You don't get it We have to work harder to outshine Skype . : ) Its not like ford and GM. They do too very different things...we don't have to outshine them we already do. On Sat, 26 Mar 2005, Stephen wrote: Hi All, Thanks for all the comments and opinions. I think in terms of features and

[Asterisk-Users] ASTERISK AT HOME USERS -- READ

2005-03-29 Thread Mike
Please use the Asterisk at home forms at http://sourceforge.net/forum/?group_id=123387 For you asterisk at home help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] using amp with asterisk?

2005-03-30 Thread Mike
Please seek help from the amp people as this list is for asterisk. On Wed, 30 Mar 2005, hank smith wrote: hello I have asterisk 1.0 running on fedora core3 and amp version 1.06 I think is the version its the version down below the current release, I have fwd working threw iax on outbound calls

Re: [Asterisk-Users] Starting with Asterisk-SIP

2005-04-02 Thread Mike
On Sat, 2 Apr 2005, ruben cuevas rumin wrote: Hi all, I'm a Telecomunication Engeenering student. I have to develop a VoIP apliccation using SIP protocol. I have to develop the SIP Server, and the SIP clients. I think I can use Asterisk for this issue. I have installed it and I have run it, but I

[Asterisk-Users] Meetme conf and Shoutcast

2005-01-16 Thread Mike
We would like to know if there is a way to broadcast (in realtime) a conferance. We hold large phone conferances and would like to know if we could have some of our users listen over a streaming services. Formats we have looked at include: Shoutcast,Real Networks,QuickTime, and dare I say

RE: [Asterisk-Users] VOIP phone suppliers in the UK?

2003-05-28 Thread mike
company Mike -Original Message- From: nathan [mailto:[EMAIL PROTECTED] Sent: 28 May 2003 11:18 To: asterisk-users Subject: [Asterisk-Users] VOIP phone suppliers in the UK? Hi All, Can anyone recommend a supplier and/or a particular a model of SIP voip phones in the UK? I don't

RE: [Asterisk-Users] Telephone Tree

2003-06-12 Thread mike
. Their incessant complaint (the police, that is) is that it's too expensive to roll out much beyond a trial system. Driving the hardware cost down would just leave them their revenue costs to whinge about. Mike Pellatt ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] Struggling with ISDN4Linux and Asterisk config

2004-04-07 Thread mike
' and nothing in the logs. Can someone help me with this? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[asterisk-users] Problem with the dial command

2007-08-01 Thread Mike
not getting either a ring or a no route to destination error. It's as if Asterisk is trying to reach the phone for the full 15 seconds, and only then giving up. My tests are done with a Polycom 650 phone, if that matters (I doubt it does). I've seem the same behavior on Polycom 501 and 320. Mike

Re: [asterisk-users] Problem with the dial command

2007-08-01 Thread Mike
? Is this any other obvious option that escapes me? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Wednesday, August 01, 2007 14:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem

Re: [asterisk-users] Problem with the dial command

2007-08-01 Thread Mike
available again after the configured number of milliseconds? Or will it be considered unreachable until the next register attempt by the device? Regards, Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Cennami Sent: Wednesday, August 01, 2007 17:56

[asterisk-users] Linksys 224P switch and Polycom PoE phones

2007-08-05 Thread Mike
have PoE enabled. From the switch to our test phone, we have a typical blue RJ-45 cable, going into the special PoE-RJ45 cable Polycom provides with the 501. And then that cable into the phone. What the heck could be wrong in such a simple setup? Mike

[asterisk-users] Buddy watch and the hint priority - brain teaser

2007-08-08 Thread Mike
(hint_reg=${EXTEN}-reg} exten = _XXX,hint,SIP/${hint_reg} exten = _XXX,SIP/${EXTEN}-reg} Or, even easier (if it can even be done) is write a function: exten = _XXX,hint,SIP/ReturnCorrectRegistration() What's the best way to approach my problem? Mike

[asterisk-users] How to write a function with a return value in Asterisk

2007-08-08 Thread Mike
be done in the same Asterisk priority. See my previous email for background (Buddy watch and the hint priority - brain teaser). Any help is extremely appreciated. Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users

Re: [asterisk-users] How to write a function with a return value inAsterisk

2007-08-08 Thread Mike
(SIP/${A}) ; I need to know ${A} first, but I can't know before this line is called (it's very DB driven). What can I do? Am I dead in the water here? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, August 08, 2007

Re: [asterisk-users] How to write a function with a return value in Asterisk

2007-08-08 Thread Mike
)})) , what I don't know is how to actually write the function with a return value (and Googling this doesn't get me any relevant result, apparently). I'd be most thankful for some link to a page that shows how to write such a function in Asterisk. Mike -Original Message- From: [EMAIL PROTECTED

Re: [asterisk-users] How to write a function with a return value in Asterisk

2007-08-08 Thread Mike
)}) In the hope of getting to see Noop(Hello World) in my CLI, I get the following Asterisk error: Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not registered AGI certainly seems registered as it worked in the first case. Again, something obvious I missed? Thank you, Mike

Re: [asterisk-users] How to write a function with a return value inAsterisk

2007-08-08 Thread Mike
then actually integrating that code in larger project...unfortunately. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Wednesday, August 08, 2007 14:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users

[asterisk-users] Using CURL

2007-08-08 Thread Mike
? Mike ___ --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] Paging Application - Polycom 601

2007-08-09 Thread Mike
to firmware 2.x and get whatever benefits you can get from that. Regards, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Thursday, August 09, 2007 10:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re

[asterisk-users] The quest for making hint more flexible continues - using Realtime now

2007-08-09 Thread Mike
:11187 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for that extension Wellthere is! Is there any way I can do this? Mike ___ --Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] The quest for making hint more flexible continues - using Realtime now

2007-08-09 Thread Mike
I feared so, but I have already started working on this. Thanks for the confirmation. Too bad, the rest of my design was relatively elegant (IMO) and easily to modify. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent

Re: [asterisk-users] The quest for making hint more flexiblecontinues - using Realtime now

2007-08-09 Thread Mike
subscribecontext (one word) is another attribute of a peer (sip.conf). I am using it as part of a MYSQL table that holds all my sip registrations, and that works fine. I did have to add the column, since it wasn't part of the table construct that can be found on the wiki. Mike

Re: [asterisk-users] Forced Ping or re-registration process for SIPdevices or accounts/lines

2007-08-09 Thread Mike
Possibly NAT related issues. Try to add the line qualify=yes to your SIP peer/friend/user. I just discovered that, wonderful little gizmo. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Lengua Sent: Thursday, August 09, 2007 16:13 To: asterisk-users

[asterisk-users] FW: Can you reload only one conf file?

2007-08-09 Thread Mike
In the interest of making things cleaner, I'd like to know if I can just reload one single conf file. Let's say I have two files, extensions.conf which includes small_file.conf. I only want small_file.conf reloaded, not the main file. Is this at all possible? Mike

Re: [asterisk-users] FW: Can you reload only one conf file?

2007-08-10 Thread Mike
The thing is that I make them automagically reload from outside Asterisk (by calling asterisk -rx extensions reload) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Friday, August 10, 2007 10:32 To: Asterisk Users Mailing List

Re: [asterisk-users] FW: Can you reload only one conf file?

2007-08-10 Thread Mike
that mattered as opposed to the whole thing. For all I know, this could be triggered while I am coding some new thing and could screw up my dialplan. But I guess I won't be doing this. Regards, Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent

Re: [asterisk-users] FW: Can you reload only one conf file?

2007-08-10 Thread Mike
this? Because the %*$%/$ hint fonctionnality can't accommodate variables fetched from a DB like the rest of my dialplan. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Friday, August 10, 2007 12:11 To: Asterisk Users Mailing List - Non

[asterisk-users] Polycom question - removing a soft key functionality

2007-08-10 Thread Mike
that they can see status by looking at the line icon, this will only confuse them). Second question, can you set up the phone so that this status, which is shown in the line icons, is also shown in the contact directory? Regards, Mike ___ --Bandwidth

Re: [asterisk-users] FW: Can you reload only one conf file?

2007-08-10 Thread Mike
that and specified extension reload foo.conf Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, August 10, 2007 09:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: Can you reload only one

Re: [asterisk-users] Polycom question - removing a softkeyfunctionality

2007-08-10 Thread Mike
of very obvious typos/spelling mistakes. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Friday, August 10, 2007 10:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom question - removing

[asterisk-users] flash zap FXO port from SIP device (SPA-2002) using RFC2833 or SIP INFO

2007-08-19 Thread Mike
debug and doesn't work. using RFC2833 (AVT) and application/dtmf-relay does the same as above. Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] Trying to use Set Group correctly

2007-08-29 Thread Mike
is not that I want. How can I make sure that only the external leg is counted? Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?

2007-09-25 Thread Mike
I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?

2007-09-26 Thread Mike
I use a 650, so YMMV, but it's working with mine. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, September 26, 2007 01:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes

[asterisk-users] DID number

2008-03-02 Thread Mike
hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

[asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

2008-04-09 Thread Mike
disable colorization from the command line, and I did try using nocolor=no in the config files. No luck. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

2008-04-09 Thread Mike
? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez Sent: Wednesday, April 09, 2008 19:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[asterisk-users] problem with Asterisk 1.4.19 - accountcode dissapearing

2008-04-15 Thread Mike
Hi, I have a big issue during transfers (using Polycom phones, but I don't think that's relevent) with Asterisk 1.14.19. Basically, the value contained in ${CDR(accountcode)} dissapears. Here is the relevant code snippet: -- exten =

Re: [asterisk-users] problem with Asterisk 1.4.19 -accountcode dissapearing

2008-04-16 Thread Mike
Thanks, that`s what I ended up doing. Still, it doesn't seem to be WAD, since the CDR(accountcode) is correct and suddently dissapears. Is this a bug (I was looking through the bug system and couldnt match this with a bug, but then again I am not a developer) or is it really WAD? Mike

[asterisk-users] Using Chanspy

2008-04-16 Thread Mike
Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another

[asterisk-users] Problem with hints (1.4.19)

2008-04-16 Thread Mike
Hi, (me again, my upgrade to 1.4 is more painful then I imagined it would be). I just noticed that the command show hints shows all hints correctly, but none of them ever are InUse (even if I use a line and dial out) like I used to on 1.2. Can`t find a bug in the bug tracking system, is

Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!

2008-04-17 Thread Mike
. And the good thing is you often do not have to do anything but set the upload bandwidth (yes there is an automatic mode, but it's not that great). Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, April 17, 2008 10

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

2008-04-17 Thread Mike
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The other time, it crashes Asterisk. Using 1.4.19 too. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Rawlings Sent: Thursday, April 17, 2008 14:10 To: Asterisk Users

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

2008-04-17 Thread Mike
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Thursday, April 17, 2008 17:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19 Mike wrote: My own

Re: [asterisk-users] Newbie Polycom: Subscription/Presence Problem

2008-04-18 Thread Mike
when the phone is ringing/on a call. Asterisk doesn't support all those fancy status that you can select from the phone. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

2008-04-18 Thread Mike
Anthony, What bug report ID# would that be? Not being a dev I find it hard to know which of the 4 chanspy bug I need a patch for, since none of them seem to refer to a 1.4.19 bug. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony

Re: [asterisk-users] Newbie Polycom: Subscription/Presence Problem

2008-04-18 Thread Mike
config looks good. Regards, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Friday, April 18, 2008 08:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie Polycom

[asterisk-users] Question on groups

2008-04-18 Thread Mike
(accountcode)}) And the CLI shows : -- Executing [EMAIL PROTECTED]:2] Set(SIP/00041234432-1-b7d4b908, GROUP=internal-5149070849) in new stack Which seems right. But it never shows up in the CLI when checking for group channels. Any clue? Did something major change between 1.2 and 1.4? Mike

Re: [asterisk-users] Question on groups

2008-04-18 Thread Mike
Wow that was easy. Thanks! My wrong syntax must have worked in 1.2 by pure chance, or I must have erased the brackets by mistake while I was tweaking the config files following the upgrade. Thanks alot, you saved me alot of time and grief. Regards, Mike _ From: [EMAIL PROTECTED

[asterisk-users] Problem with transfer (and asterisk -r)

2008-05-05 Thread Mike
Hi, I used to have ## configured in asterisk 1.2 for blindxfer. Now, when I press ## I hear it on the other end instead of initiating a transfer. What has change and how can I go back to the old behavior? I kept the same feature.conf file with these lines: [featuremap] blindxfer = ##

Re: [asterisk-users] [OT] wireless headphone that can answer a call?

2008-05-05 Thread Mike
If you want to avoid a mecanical lifter, the only option I know of is a Jabra GN9350 with a Polycom EHS (electronic hookswitch) cable. It came out only a month ago I believe. I use one, and wouldn't give it up for the world. Regards, Mike -Original Message- From: [EMAIL PROTECTED

[asterisk-users] Using multiple variables in SIP.CONF setvar

2008-05-12 Thread Mike
Hi, What is the syntax to set more than one variable in the SIP.conf file for a particular sip peer? (using the setvar line) Regards, Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

[asterisk-users] Problem with Polycom forwarding

2008-05-20 Thread Mike
Hi, I am having trouble with Polycom forwards and Asterisk. Basically, I have no clue on how to force callerid or even custom variables (set using SetVar in the sip.conf file) on the transfered call. For example, I set a variable called var_a to foo. When the call comes in, the variable

Re: [asterisk-users] Problem with Polycom forwarding

2008-05-20 Thread Mike
using realtime SIP entries) and not in the diaplan per say. My setvar column is this: internal_callerid=blabla 123;did=551234 I tried adding underscores before did (as in: internal_callerid=blabla 123;__did=551234) but that didn't help. Mike

Re: [asterisk-users] Problem with Polycom forwarding

2008-05-20 Thread Mike
Did you try _var_a? Iirc you need to prepend it with an underscore to make the variable persistent. Forget my previous email, it didn't quite work that simply but I tweaked my dialplan and you had the right solution. Thank you, Mike

Re: [asterisk-users] Trouble with Polycom phones

2008-06-05 Thread Mike
Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running

[asterisk-users] Asterisk : using setvar with IP Realtime and variable inheritance

2008-06-09 Thread Mike
Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar

Re: [asterisk-users] Asterisk : using setvar with IP Realtime and variable inheritance

2008-06-10 Thread Mike
To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk : using setvar with IP Realtime and variable inheritance Mike wrote: If I hardcode this value in my dialplan using two underscores before it (i.e Setvar(__did=551234) ) this works. But I can't

[asterisk-users] Losing CDR(accountcode)

2008-06-11 Thread Mike
Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick

Re: [asterisk-users] Losing CDR(accountcode)

2008-06-11 Thread Mike
at 2:21 PM, Mike [EMAIL PROTECTED] wrote: Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL

[asterisk-users] Dialing vs forward - was RE: Asterisk : using setvar with IP Realtime and variable inheritance

2008-06-12 Thread Mike
See, to get back to your answer, this is what I`m not understanding: Again, this works fine. The problem is when I forward my calls to another outside line (using Polyocm phones), and need to know the ${did} value at that point. It's empty. Right, so the call path is: Provider --

[asterisk-users] Problem with realtime?

2008-06-17 Thread Mike
I get that a lot since moving to 1.4.21 (from 1.4.18 or something). [Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Question 1: what debug file should I be looking at? Mick

Re: [asterisk-users] Problem with realtime?

2008-06-17 Thread Mike
Just an addition: that happens big time when I do a sip reload from the CLI I know this should help me already, but it doesn`tÂ… From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, June 17, 2008 09:23 To: 'Asterisk Users Mailing List - Non-Commercial

[asterisk-users] TRANSFER_CONTEXT ignored?

2008-06-18 Thread Mike
Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is internal. Outgoing calls go through that context (all good). When I get an incoming call which I

[asterisk-users] Question on screening calls / Question about the Dial g option

2008-10-07 Thread Mike
the call initiated using the Dial g option is hung up ? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] Question on screening calls / Question about the Dial g option

2008-10-07 Thread Mike
Doug, Thanks for the quick answer. How does that help me though, since this is a per channel variable and not a global variable? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, October 07, 2008 16:54 To: Asterisk Users

[asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

2008-10-09 Thread Mike
. I've played around with the kewlstart and loop-start setting but without knowing what the line is going to do, it's difficult to know how to configure Asterisk. Does anyone have any experience of Telewest? Thanks, Mike. ___ -- Bandwidth and Colocation

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

2008-10-09 Thread Mike
On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote: Mike, Can you tell us : - asterisk version - zaptel version When you call over this line, when you hangup did you hear an busy tone ? or any class tone ? To do this test connect your lines to analog phone and make a call

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

2008-10-13 Thread Mike
On Sun, Oct 12, 2008 at 11:18:40AM +0100, Gordon Henderson wrote: On Thu, 9 Oct 2008, Mike wrote: I'm guessing this lamp is on an ordinary analogue phone you have? Yeah, this is a bog standard 9 quid analogue phone. OK. A bit convoluted this as I'm not local to the PBX, but an IAX trunk

[asterisk-users] asterisk setup

2008-10-20 Thread Mike
in advance, Mike ___ -- 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] asterisk setup

2008-10-20 Thread Mike
What country are you in? This is a truly global marketplace and mailing list. We have people from the UK, Ireland, Oztrailia, New Zealand, Bolivia, Russia, China, India, Argentina, etc. All over the world, really. Saying what country you need the DID/DDI in will narrow it down somewhat. I am

[asterisk-users] TDM400 with FXS some handsets not ringing

2008-11-05 Thread Mike
can do to make my other phones ring? Thanks, Mike. ___ -- 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] Polycom 501 issue withlatest firmware: sluggishkeys - new info

2007-04-16 Thread Mike
(new in 2.x) Let me know if that worked. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, April 16, 2007 09:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue

[asterisk-users] Best Wireless bridge for Polycoms

2007-04-27 Thread Mike
for that matter) into a WiFi phone? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Best Wireless bridge for Polycoms

2007-04-27 Thread Mike
, where do I find an adapter for NA power that turns into 2V 5A DC current? Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, April 27, 2007 13:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best

[asterisk-users] Headset for Polycom

2007-05-04 Thread Mike
Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike

[asterisk-users] Dealing with 2 SIP providers

2007-05-11 Thread Mike
in the same SIP entry? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Dealing with 2 SIP providers

2007-05-11 Thread Mike
Yeah ok. That doesn't help. What I mean is I want a call to go out on ProviderA, UNLESS it's down and then go to ProviderB. I want it to ring 30 seconds and then Hangup if nobody has answers. I DON'T want to dial both, only one or the other. Mike -Original Message- From: [EMAIL

RE: [asterisk-users] Dealing with 2 SIP providers

2007-05-13 Thread Mike
of programming with every line numbered like BASIC Can you easily mix and match AEL and standard Asterisk (i.e. my old code with new code I would put in?) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Friday, May 11, 2007 22:22

[asterisk-users] Where to find Polycom firmware with 330/320 support?

2007-06-05 Thread Mike
Hi, I just got a Polycom 330 and, of course, I don't have the firmware and sip.cfg files to provision it. Where can I get those? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Do I understand GROUPs correctly?

2007-02-27 Thread Mike
the number doesn't increased as planned, it stays at 1 channel used. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

RE: [asterisk-users] Do I understand GROUPs correctly?

2007-02-27 Thread Mike
) twice. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, February 27, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Do I understand GROUPs correctly? Mike wrote: Hi

RE: [asterisk-users] Do I understand GROUPs correctly?

2007-02-27 Thread Mike
into a Set(group) again to increment channel before dialing a cell phone using a dial(cellphone#) cmd. If that doesn't work, how do I accomplish the same kind of thing elegantly? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent

RE: [asterisk-users] Do I understand GROUPs correctly?

2007-02-28 Thread Mike
Thank you, that is exactly what I needed. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, February 27, 2007 11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do I understand

RE: [asterisk-users] RE: Polycom reject button

2007-03-02 Thread Mike
Jason, If you do test if JR's tip works, please share your finding with us. I am interested in this as well. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Thursday, March 01, 2007 21:11 To: asterisk-users@lists.digium.com

[asterisk-users] Rebooting ALL polycom phones

2007-03-12 Thread Mike
polycom phones that I wanted to reboot)? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Rebooting ALL polycom phones

2007-03-12 Thread Mike
Thanks Dave, good info! And thanks to those who confirmed I needed to write a script because there were no built in functions, I appreciate that info too. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Monday, March 12, 2007

[asterisk-users] CDR and CallerID

2007-03-13 Thread Mike
values must be identical. Is there any way to change that? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PoE - IEEE 802.3af

2007-03-28 Thread Mike
! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth

RE: [asterisk-users] PoE - IEEE 802.3af

2007-03-28 Thread Mike
Thanks for all the replies, this definitely helps me! Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, March 28, 2007 12:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE

[asterisk-users] Polycom 601 message waiting indicator

2007-04-05 Thread Mike
a message waiting? Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] What is this error message? (check_auth: stale nonce received from ...)

2007-04-05 Thread Mike
I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED] I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike

RE: [asterisk-users] Polycom 601 message waiting indicator

2007-04-05 Thread Mike
Thanks David and Chris, appreciate the response Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, April 05, 2007 11:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom

[asterisk-users] Polycom 330/320

2007-04-09 Thread Mike
useful. Also, I read that the phone offers TLS security. What does that mean? I understand Asterisk does not, but is this something that could be possible with futur asterisk developement? Mike ___ --Bandwidth and Colocation provided by Easynews.com

RE: [asterisk-users] Polycom 330/320

2007-04-09 Thread Mike
Ah, thanks. I didn't realize this. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Monday, April 09, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 330/320

[asterisk-users] Reverse-ATA : Using PSTN lines to connect to Asterisk

2007-04-10 Thread Mike
in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

RE: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk

2007-04-10 Thread Mike
Probably, if I only needed one FXO. What is the customer has 4 channels (PSTN lines)? Don't I need 4 FXO? And, about the Sipura, it looks like it would do what I want, but it only has one FXO, limiting it's usefulness. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

RE: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk

2007-04-10 Thread Mike
Thanks Alex, That was my original thought, to just buy a TDM400 from Digium and put in as many FXO as I wanted, but I liked having the ease of just buying something off the shelf, even if it meant paying a little more. But it looks like I won't have much of a choice. Mike -Original Message

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