On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote:
> 25.12.2014 15:46, Anthony Messina пишет:
> On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote:
> I want to change call files, which has caller id in them, to call
> originate from dial plan.
> But I
On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote:
> I want to change call files, which has caller id in them, to call
> originate from dial plan.
> But I don't see such parameter here
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate
>
> How can I pass
On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote:
> Is there any reason why ODBC voicemail storage requires varchar for most
> fields? For example, is there anything stopping me using a BIGINT or
> similar for origtime or INT for duration?
It may cause you trouble when using PostgreSQL:
On Wednesday, September 17, 2014 04:35:14 PM Russ Meyerriecks wrote:
> Patch for this has been committed to master here:
> http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=b9a8000bbd1
> b6120f22627c105a2c2194dcc793d
>
> I expect to release a v2.10.1 for this soon.
> Thanks for the
On Saturday, September 13, 2014 03:15:57 PM sean darcy wrote:
> On 09/13/2014 01:52 PM, sean darcy wrote:
> > On 09/13/2014 12:09 PM, sean darcy wrote:
> >> On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against
> >> it. dahdi show channels works fine, but when I try to place a ca
On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote:
> Hi Anthony,
>
> That script does not work. My guess is that it is related to the way
> asterisk interacts with CentOS environment.
>
> Best Regards,
> Paul Greenberg, Esq.
>
> On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez
On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote:
> I installed CentOS 7 on a spare server along with all our Asterisk
> configuration system and the only thing that failed is the asterisk
> startup script included in the asterisk tarball. I guess because the
> startup system
On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote:
> On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote:
> > On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote:
> > > On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
> >
On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote:
> On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote:
> > On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
> > > Unfortunately, after
> > >
> > >
> > >
> > &g
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
> Unfortunately, after
>
> http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1
> 2cc0661f3810ef47ad33206b2e398
>
> I am unable to build DAHDI-Linux in a mock chroot for packaging
> purposes
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398
I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I
believe this is related to the Makefile calling install_firmware with only 2
args, where in
On Wednesday, June 19, 2013 11:11:17 AM Matthew J. Roth wrote:
> Eloi Bail wrote:
> > I am trying to enable SIP SIMPLE communication in my test environment.
I use the following which semi-enables message broadcasting to multiple
devices so a user who receives a message can reply from any of the d
On Monday, January 28, 2013 08:06:38 AM Anthony Messina wrote:
> On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote:
> > Who do I need to poke to get the yum repository / RPM files updated? The
> > dahdi RPMs are not up to date with the CentOS kernel versions any more,
On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote:
> Who do I need to poke to get the yum repository / RPM files updated? The
> dahdi RPMs are not up to date with the CentOS kernel versions any more,
> it's making doing an installation a bit tricky due to dependancies, I'd
> rather not rol
On Saturday, November 03, 2012 09:32:37 PM Eric Smith wrote:
> How would I apply the patch included in the above url?
>
> [eric@pepper ~/src/asterisk-complete/asterisk/dahdi/2.6.1+2.6.1] $ patch
> Perhaps you should have used the -p or --strip option?
You'll need to use.the -p or --s
On Friday, August 31, 2012 06:48:46 PM Noah Engelberth wrote:
> I’m trying to set up a way that our users can send an XMPP message to
> Asterisk (unsolicited) to request information, such as voicemail status or
> the like. No matter what I set for the dialplan, I’m only seeing Asterisk
> execute t
On 12/02/2011 11:37 AM, Anthony Messina wrote:
> I've just connected my new Android (Motorola RAZR) phone to Asterisk
> using CSipSimple and have discovered that on any call between CSipSimple
> and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
> he
I've just connected my new Android (Motorola RAZR) phone to Asterisk
using CSipSimple and have discovered that on any call between CSipSimple
and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
hear a rhythmic tapping as if my voice stream is being chopped up in
equal parts abou
On 08/20/2011 07:00 AM, Tim King wrote:
> exten => h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php
do you need the -f option to php?
exten => h,n,System(/usr/bin/php -f /var/lib/asterisk/bin/faxnotify.php
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89
On 08/06/2011 09:49 PM, Bruce Ferrell wrote:
> Errors follow:
http://lists.digium.com/pipermail/asterisk-users/2011-July/264993.html
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
signature.asc
Description: OpenPGP di
On 06/08/2011 01:09 AM, Paddy Grice wrote:
> Hi All
>
> I am looking for a small scale Email to fax solution
>
> Searches seem to throw up
>
> AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to
> http://www.noojee.com.au/products/noojee-fax/fax-overview/
> email12fax ht
On 05/24/2011 01:07 PM, e...@erols.com wrote:
> I have tried faxing to the DID from 2 different fax machines connected to
> different POTS lines. One fax machine is a Xerox Workcentre, and the other
> is a Brother Intellifax. Can you provide some more information about your
> setup? If you wo
On 05/20/2011 01:20 PM, e...@erols.com wrote:
> #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to
> receive faxes via T.38. Sending faxes is not a requirement. Does anyone
> have a working asterisk 1.8.4 configuration and ITSP provider that they can
> recommend? We hav
On 04/27/2011 02:06 PM, satish patel wrote:
> Which echo cancellation is good between OSLEC and MG2. Dahdi by default use
> MG2 echo cancellation on channel. If i want to use OSLEC then what should i
> need to do ? Do i need to recompile dahdi with OSLEC ?
Yes, you would need to compile the OSL
After upgrading to 1.8.3.2 today, I notice that my Aastra 480i SIP
phones no longer initiate hold music when a call is placed on hold.
I seem to be having the same issue as the person here:
http://forums.digium.com/viewtopic.php?f=1&t=77553
Has anyone else run into this issue?
--
Anthony - http:
On 03/18/2011 06:48 PM, Steve Edwards wrote:
> On Sat, 19 Mar 2011, Gilles wrote:
>
>> Thanks but for some reason, after calling out through a call file,
>> Asterisk doesn't jump to it although the callee hangs up while
>> Asterisk is still playing:
>
> Somehow, I'm guessing that 'failed' means t
On 03/18/2011 05:43 PM, Gilles wrote:
> On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
> wrote:
>> On Fri, 18 Mar 2011, Danny Nicholas wrote:
>>> I believe you will achieve the desired result by replacing ${REASON}
>>> with ${HANGUP_CAUSE}.
>>
>> REASON is documented as being valid in th
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote:
> http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script)
>
> Is your .agi and .git the same script? I do not have a git client on
> this host to see for myself.
I keep the AGI in Git as a version control system. But, you can vie
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote:
> Does the AGI have to be used? In this example
> http://www.davidvossel.com/?p=28 I see mention of a script, but not in
> this one:
> http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
>
> I believe I missing the conne
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
> Evening,
>
> Has anyone seen a how-to on getting Asterisk to work with Google Talk
> and Google Voice?
>
> Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
On Saturday, August 21, 2010 02:19:00 pm Steve Edwards wrote:
> Wow. I thought I knew a bit about bash.
>
> I made notes on 19* different lines I have no clue what they do. It's
> going to take me hours to figure these out so I can add them to my
> repertoire.
>
> *) I'm sure there's more nugge
On Friday, August 20, 2010 10:35:10 am Olivier wrote:
> Yes, adding this kind of link should do it but I'm looking for a solution
> which automatically insert whatever is needed to launch a call.
wouldn't it be difficult to know exactly which applications are available on
the system which has the
On Wednesday, August 11, 2010 11:08:37 am Tino wrote:
> #!/bin/bash -x
> T="$agi_uniqueid"
>
> I want to save value of 'agi_uniqueid' channel variable into a variable
> called 'T' in my script
When executing and AGI from the dialplan, it will dump out it's variables
immediately, so you need to t
On Monday, July 26, 2010 09:55:38 am Tzafrir Cohen wrote:
> > I suppose I should make a list of known good packages, and put it on
> > that FAQ page.
> >
> >
> >
> > GIMP is useless for FAX. Not only does it get the shape of the images
> > wrong, it can only display the first page of a FAX. I am
On Monday, July 19, 2010 01:03:57 am Peter Childs wrote:
> One of the problems with Distinctive Ring tones is that its not
> consistent, between different phones so if you have a mix of phone
> types you have a problem.
Agreed. I only mentioned what I did since I, along with the OP use Aastra
ph
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote:
> Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
> how can one receive distinctive ring tones for INTERNAL calls ONLY?
Using Aastra 4801 CT phones...
[external-context]
; Calls entering from outside the system
e
On Tuesday 11 May 2010 01:25:30 pm Tim Nelson wrote:
> I have a handful of Asterisk 1.4.x installations where users dial 'outbound
> calls' to the PSTN even though the destination is on the same Asterisk box
> or on another Asterisk box on the same network. Instead of paying twice
> for the call to
On Sunday 07 March 2010 05:10:02 pm sean darcy wrote:
> Good. Glad it we figured it out. BTW, is your src.rpm for dahdi-linux
> available?
>
> sean
Here you go. -A
http://messinet.com/pub/fedora/linux/updates/12/SRPMS/dahdi-
linux-2.2.1-2.fc12.src.rpm
--
Anthony - http://messinet.com - http:/
On Sunday 07 March 2010 09:16:55 am sean darcy wrote:
> Well, I've figured it out, at least for me.
>
> Another driver was grabbing the TDM400P: netjet.
>
> added netjet to /etc/modprobe.d/blacklist.conf.
>
> I think you can do this by:
>
> cat /lib/modules/`uname -r`/modules.pcimap | grep 00e1
On Saturday 06 March 2010 09:18:13 pm sean darcy wrote:
> I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
> installed dahdi-2.2.1.
>
> kernel modules loaded.
> lsmod | grep wctdm
> wctdm 37233 0
> dahdi 194985 1 wctdm
>
> lsmod | grep da
On Thursday 04 February 2010 23:22:27 Alex Samad wrote:
> What I have seen on my asterisk box when I had a up/down adsl line was
> that the asterisk box couldn't do dns resolution and would hang( well no
> other internal calls could be made, seemed like some sort of semaphore
> was stuck) when the
On Tuesday 05 January 2010 17:30:31 Joseph L. Casale wrote:
> Just on my way to work on this server now, this would be great! That
> way I don't have to work all night:) Does the atrpms ones finally do oslec?
I don't use them myself, but I was thinking that the RHEL5 spec files might be
another p
On Tuesday 05 January 2010 17:09:32 Joseph L. Casale wrote:
> From what I can tell so far, I can continue to use his user tools unchanged
> but I need to apply this patch to the tar file in the
> dahdi-linux-2.2.0.2-1_centos5.src.rpm and rebuild it, but that ,
> `dahdi-linux` pulls in
>
atrpms.n
On Tuesday 05 January 2010 12:21:15 Joseph L. Casale wrote:
> So this script builds them with the dahdi-tools-libs package requirement, I
> thought the fedora spec built all of these? Any idea?
>
Fedora packages the dahdi-tools* suff, but can't include the kernel modules.
I did not realize you w
On Monday 04 January 2010 07:16:49 Joseph L. Casale wrote:
> Looking at the source in the rpms from the asterisk package site
> appears that oslec is not built and enabled for the kmod rpms.
>
> Anyone know an existing repo or have direction on how to enable
> this to built for those rpms?
>
I b
original message-
From: "mickael ropars" mrop...@gmail.com To: "Asterisk Users Mailing List -
Non-Commercial Discussion" asterisk-users@lists.digium.com Date: Fri, 27 Nov
2009 11:18:30 +0100
-
> Hi Michal,
>
>
On Monday 05 October 2009 12:33:47 Danny Nicholas wrote:
> What are the limitations of ActionID? In all of the examples I see, it is
> usually 1 or some integer. Can it be a timestamp like uniqueid?
I use AMI as part of an external bash application and I usually specify the
ActionID to the some
On Wednesday 23 September 2009 01:44:31 sean darcy wrote:
> Does anyone use SendFax for analog faxing?
>
Yes. I have two contexts as follows:
[outbound]
exten => _X.,1,Dial(DAHDI/G2/${EXTEN})
[sendfax]
exten => s,1,SendFAX(${FAXFILE})
exten => h,n,Hangup()
When I want to send a fax, I initi
On Monday 07 September 2009 16:27:30 Carlos Chavez wrote:
> It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT)
> have a problem with the new SIP implementation in Asterisk 1.6.X that makes
> them unable to dial. They can receive calls but when you attempt to dial
> the phone
On Monday 07 September 2009 13:40:16 jonas kellens wrote:
> [applicationmap]
>
> opnemencallee =>
> #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
=>
,[/],[,[,MOH_Class]]
it looks like "/var/samba/profiles/jonaskl/recording" is in the spot for
"[,MOH_Class]"
--
Anthony -
On Wednesday 12 August 2009 08:30:33 am harry R wrote:
> Or maybe can suggest another CDR GUI ?
i began work on this a while ago...
http://messinet.com/trac/webcdr+/
it's what i use now, though i'd like to add more features, etc.
--
Anthony - http://messinet.com - http://messinet.com/~amessina/g
Recently, I've been having issues with the URIs returned from e.164.org and
toll free calls. It seems that the URIs that are returned from ENUMQUERY and
ENUMRESULT are no longer the proper numbering schemes that the poviders use.
I've been using the following [enum] template in my outbound route
On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote:
> -> To resolve this i tried to remove all keys in all servers and once
> again created and
>distributed the loaded in each system with "keys init" command but
> stilll i am
>getting the same error
>
>
>
> can anybody help me out
On Friday 29 May 2009 11:20:31 am David Backeberg wrote:
> On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA
>
> wrote:
> > i cannot originate call from AMI interface here are my Originate action
> > Packet
> > Channel: SIP/111
> > where 111 Is my SIP phone number which registered with my asterisk
original message-
From: "Jimmy Godbout" s...@inbox.com
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Date: Mon, 25 May 2009 18:01:11 -0800
-
> Check on www.localca
On Friday 15 May 2009 03:49:05 pm Hristo Benev wrote:
> I came up to this solution, but is there a way to change the AGISTATUS
> variable to FAILURE -> We have it always SUCCESS
if the script you use exits successfully (without an error), AGISTATUS will
always be SUCCESS even if it didn't do what
On Thursday 16 April 2009 09:52:45 Danny Nicholas wrote:
> I've got 1.4.21.2 using Polycom 501 phones and
> Zap lines. Most of my calls come in and go out fine with the exception of
> Mechanized answering devices. When I call my 401K plan (1-800-777-401K)
> the call will
On Tuesday 10 March 2009 13:32:37 Elizabeth Steinke wrote:
> Greetings!
>
> We are using cisco 7940 phone with SIP and asterisk. We would like to be
> able to have phone directories available on the phones that are sourced
> from active directory. Are their any scripts that can connect to the AD
>
On Thursday 05 March 2009 07:10:59 Kevin P. Fleming wrote:
> Peter Mueller wrote:
> > Has anybody set up such an installation and/or is OpenAIS able to
> > transfer the devstates over different subnets? Haven't found docs and
> > hints for this use case.
>
> The method OpenAIS uses to communicate b
On Friday 27 February 2009 17:02:16 Daniel Hazelbaker wrote:
> > Or, if you're using Asterisk 1.6 and looking to try something new,
> > take a look
> > at http://messinet.com/AsteriskFAXGateway
>
> I'll take a look at both packages. I hadn't given HylaFAX(+) any
> thought as when I searched in
On Friday 27 February 2009 14:03:19 Doug Lytle wrote:
> Daniel Hazelbaker wrote:
> > Specifically, I am trying to play around with setting up a fax
> > server. I can receive the fax, but sometimes the sending fax hangs up
>
> If your looking into setting up a reliable fax server and your not doing
On Friday 13 February 2009 11:39:07 Carlos Chavez wrote:
> Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
> Making calls is not a problem but when you receive a call it always
> drops at 1:45 minutes, always!
I use 1.6.0.5 with 3 Aastra 480i CT phones and have no issues
On Friday 13 February 2009 07:54:48 Jeff LaCoursiere wrote:
> Is there a Chicago area users group? If not is there any interest in
> creating one?
there is: http://groups.google.com/group/asterisk-chicago
though it's fairly inactive.
--
Anthony - http://messinet.com - http://messinet.com/~ames
On Wednesday 31 December 2008 18:07:26 Karl Fife wrote:
> Allison Smith just created a hysterical parody music on hold Parody.
> Whatever you were doing, stop, and dial this number to listen to it:
> 360-519-5689. 2 minutes.
>
> I just gave her a few ideas, but she took it and ran with it--she cho
On Tuesday 23 December 2008 05:00:10 Tzafrir Cohen wrote:
> And this is a reminder: they don't queue mail. Hence if they fail to
> deliver once, the mail is lost. May not be the best idea for sending
> mail over the internet.
This is a great reminder as many systems use graylisting. Emails to tho
On Tuesday 23 December 2008 01:05:40 jordan pan wrote:
> Hi everyone,
>
> when i use the automated dial out,I found that once the zap answerd,the
> contex will be exectued, but i don't hope do it ,i hope when extern phone
> answered ,then ,the context will be exectued.
> Anyone can help me solv
On Friday 19 December 2008 20:24:11 sean darcy wrote:
> Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
>
> [incoming-fax]
> exten =>
> s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0
>${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif)
> e
On Saturday 13 December 2008 11:34:07 Loic Didelot wrote:
> Hello,
> which package are you using to get the application ReceiveFAX under
> asterisk 1.4?
since this is still in development, for me, i'm working with asterisk 1.6.
this package is mostly independent of the fax system within asterisk
On Friday 12 December 2008 15:41:54 Mark Michelson wrote:
> would result in a variable called FOO being set to the value
> "hello,BAR=world". The MSet application was added to facilitate being able
> to set multiple variables in a single application call. If using MSet, the
> above would instead re
On Friday 12 December 2008 12:08:55 sean darcy wrote:
> I just want to pdf and email faxes coming in over pstn on a TDM400P.
>
> Outgoing faxes would just go out over pstn, not through asterisk.
>
> All the voipinfo , etc, howto's are quite complicated. And most use
> third party apps like Hylafax.
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:
> Hi guys,
>
> I have 1 zap channel in my house shared among couple people. If someone
> dials 911, I want that zap channel to be disconnected right away to make
> way for the 911 call.
>
> I dug through voip-info.org and didn't find much.
>
On Thursday 27 November 2008 05:03:00 Olivier wrote:
> Hi,
>
> Do you have any example showing how to use SendFAX ?
> I can see several examples of ReceiveFAX but not a single one showing
> SendFAX.
i'm working on a script to incorporate e-mail <-> fax gatewaying with asterisk
using programs that
On Saturday 22 November 2008 09:10:39 am Gordon Henderson wrote:
> On Sat, 22 Nov 2008, Noah Miller wrote:
> > Hi Ken -
> >
> >> Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to
> >> receive faxes was, well, a PITA, what with having to patch the Asterisk
> >> install with var
On Thursday 30 October 2008 11:57:29 am Daniel Hazelbaker wrote:
> Is it just me or has mantis been holding onto old e-mail and finally
> sending it?
i'm getting them too. even the original "your license agreement is accepted"
email.
--
Anthony - http://messinet.com - http://messinet.com/~a
On Friday 24 October 2008 11:49:15 am Wilton Helm wrote:
> 1. Why would anyone originate a FAX via VoIP? If it has to go through a
> bunch of translation steps at both ends, it would seem better to simply
> scan the document (assuming it isn't in electronic form to begin with) and
> attach it to
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote:
> On Sun, 12 Oct 2008, sean darcy wrote:
> > Becasue of all the issues with fax over voip, we want to use pstn for
> > our fax machine, but not dedicate a line just to fax.
> >
> > I'm thinking of having asterisk answer the pstn line, ch
On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote:
> Now I have not touched any of that code, but to me, it would have been much
> simpler to change names, then change functionality later. Make DAHDI a
> drop in replacement for Zaptel, in fact, if memory serves me correctly that
> is what
On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote:
> > I think it's better to find out what is listening on port 4520.
>
> CentOS 5
> Asterisk 1.4.20
>
> Presumably my other Asterisk server is listening on 4520.
>
> The problem here is that I can change the port, and it will work...
> unt
On Saturday 30 August 2008 01:35:10 pm Karl Fife wrote:
> Indeed you're right.
> You'd have area covered by AP 'A' only, AP 'B' only and area of AB
> overlap, Picture a venn diagram:
> http://upload.wikimedia.org/wikipedia/commons/5/56/Venn-diagram-AB.png
right. it's just your inital descriptio
On Saturday 30 August 2008 11:51:49 am Karl Fife wrote:
> Let's say a 'YES' only counts if you had a bona-fide handoff. In other
> words, you began in place 'A' (within range of AP#1 but OUTSIDE the
> range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but
> completely outside the range
On Tuesday 26 August 2008 11:44:42 pm Karl Fife wrote:
> I'll be that none of the other coffee makers can handle anywhere NEAR 60
> voice channels, and don't get me started about HPEC!
>
> http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759
Good find! Does it grind it's own beans?
--
On Saturday 23 August 2008 03:56:15 am Gavin Henry wrote:
> What setup would you recommend for making VoIP calls whilst bringing
> latency down between offices at:
>
> * Edinburgh
> * Kuala Lumpur
> * Singapore
> * Tokyo
> * Seoul
> * Beijing
> * San Francisco
>
> Some of the Asia offices are > 300
On Friday 22 August 2008 07:54:43 am Anthony Messina wrote:
> On Thursday 21 August 2008 08:26:47 am Olivier wrote:
> > Hi,
> >
> > To check telco billing, I'm usinfg Asterisk-Stats from
> > http://www.areski.net/asterisk-stat-v2/about.php .
> >
> > H
On Thursday 21 August 2008 08:26:47 am Olivier wrote:
> Hi,
>
> To check telco billing, I'm usinfg Asterisk-Stats from
> http://www.areski.net/asterisk-stat-v2/about.php .
>
> How can you tweak this application to display graphics and data that use
i started working from that software to come up t
On Thursday 14 August 2008 03:09:42 pm roberto wrote:
> I'm looking for some "free" LATA X Area Code database.
>
> Anyone have any idea where can i found?
this site has lots of info: http://www.localcallingguide.com/
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote:
> For a limited time only, Messinet Secure Services (me) will be offering
> DUNDi E.164 termination to the entire +1 country code. I'd like to
> encourage more peering within the US, but peering is open to anyone
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote:
> For a limited time only, Messinet Secure Services (me) will be offering
> DUNDi E.164 termination to the entire +1 country code. I'd like to
> encourage more peering within the US, but peering is open to anyone
On Tuesday 22 July 2008 02:58:38 pm Jason Lixfeld wrote:
> I was looking for a Click to Dial/Web Dial solution and I found
> AsteriDex. I'm looking for something I can use on the road where I
> can hit an internal Click to Dial/Web Dial page from my cell, tap on a
> number and have it bridge
On Friday 09 May 2008 10:19:23 am equis software wrote:
> Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
> you think about to use Ubuntu or another distibution??
>
> Thanks
I have used Fedora 7 & 8 on both i386 & x86_64. I have used the RPMs from
atrpms.net in the past
On Thursday 10 April 2008 02:14:17 pm Steve Edwards wrote:
> > 1 - Can you really make free outgoing calls from let's say Portland OR,
> > to Frankfurt Germany?
>
> No. There is no free lunch. It takes electricity, bandwidth, and depending
> on who you want to call in Germany, termination.
though
On Thursday 20 March 2008 05:06:29 am Mian M Asif wrote:
> Hi eric,
> can you please tell me how can i save the value of EXTEN in a different
> variable before the Goto(s-${DIALSTATUS},1),
>
> thanks for you help,
>
> regards,
> Asif
>
>
> Message: 14
> Date: Wed, 19 Mar 2008 10:39:22 -0500
> From:
On Sunday 09 March 2008 09:59:32 pm Philip Prindeville wrote:
> http://bugs.digium.com/view.php?id=11969
>
> If Macro()/MacroExit() is deprecated, how does one go about achieving
> the same functionality with Gosub()/Return()?
i agree--an excellent question. Since Macro is depreciated and I am us
Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] TDM400P
> > dialout problem
> >
> >Anthony Messina wrote:
> >> Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble
> >> dialing out to the pstn. The call is init
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack
[
On Sunday 24 February 2008 02:01:10 am arkda wrote:
> Hi,
>
> I'm having difficulties with using DUNDi between two servers. If it were
> three I think I could control looping by limiting TTL, but with two I'm not
> sure how to prevent a loop causing bad things to happen. I've tried ttl=1
> but thin
On Friday 15 February 2008 01:49:46 pm Richard Lyman wrote:
> Anthony Messina wrote:
> > On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
>
> *snipped
>
> >> Priority: 1
> >> Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
> >
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
> Anthony Messina wrote:
> > Working with asterisk 1.4; using the AMI Originate command, it is
> > possible to do something like:
> >
> > Variable: CDR(accountcode)123456
> >
> > Or must the variabl
Working with asterisk 1.4; using the AMI Originate command, it is possible to
do something like:
Variable: CDR(accountcode)123456
Or must the variable names be "var[n]" where n is a number?
I'd like to set the accountcode for a Local channel that originates a call.
Thanks. -A
--
Anthony -
On Saturday 02 February 2008 02:45:09 am Alberto Pastore wrote:
> I need a bunch of them to convert some old fashioned rotary phones
> into VoIP ones (I'd like to disassemble the ATAs to remove the
> boards from the plastic case and to fit them into the phones
> after making the appropriate changes
On Thursday 31 January 2008 11:52:09 pm Ian wrote:
> Sorry for taking so long to reply,
>
> This email got lost in translation, again.
>
> Ian
>
> Ian said the following on 30-Jan-08 03:57 PM
>
> > Thaks for the speedy reply
> >
> > Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
> >> On We
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