Re: [asterisk-users] asterisk module app_konference

2012-10-03 Thread Leif Madsen
could give a better idea as to what your needs are, and why MeetMe() doesn't fulfill them? Perhaps ConfBridge() in Asterisk 10 or later would fulfill those needs? -- Leif Madsen http://www.oreilly.com/catalog/ast

Re: [asterisk-users] Case-sensitivity of Dialplan variables.

2012-10-03 Thread Leif Madsen
performed by those affected. Especially, as in the case of what Raj mentioned at the beginning of his prior email, not too many people may even be affected by this change just like he won't be. Well said. -- Leif Madsen http://www.oreilly.com/catalog/ast

Re: [asterisk-users] Case-sensitivity of Dialplan variables.

2012-10-03 Thread Leif Madsen
lable, and we're basically in the same boat as just changing it in the next major version. Consistency for the win!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Case-sensitivity of Dialplan variables.

2012-10-03 Thread Leif Madsen
based on a released major version would not be affected. +1 to case-sensitivity. It's the right way!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digit

Re: [asterisk-users] Questions on converting to ConfBridge

2012-10-03 Thread Leif Madsen
conference bridge. No need to manipulate from the dialplan anymore. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] User expected behavior of musiconhold and AGI's "stream file"

2012-09-28 Thread Leif Madsen
On 28/09/12 08:45 AM, Joshua Colp wrote: Leif Madsen wrote: I guess part of the question is; can you trigger it to be re-enabled after the stream file? Sure you can! You can use set music to start it going again as the next command. And that makes sense. I kind of knew the answer already

Re: [asterisk-users] User expected behavior of musiconhold and AGI's "stream file"

2012-09-28 Thread Leif Madsen
e. Changing this can only make it a backwards compatibility issue. Someone who has run into this and needs it to act differently will seek out the new option after reading about it in the CHANGES file. In an ideal scenario, a system upgrade should require the least amount of knob turning. -- Leif M

Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

2012-09-28 Thread Leif Madsen
s!! Oh heck ya. You can start up an Asterisk instance and just start doing things with it via your programs. That's the immense power of AMI; it's essentially the Asterisk API. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- ___

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

2012-09-28 Thread Leif Madsen
On 28/09/12 07:36 AM, Markus wrote: Am 28.09.2012 13:24, schrieb Leif Madsen: Is another channel connected to the conference receiving the DTMF? Is that what you're intending? Because from my understand that is the intention, and not simply to limit the DTMF from being in the conference i

Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

2012-09-28 Thread Leif Madsen
It uses a modified InnoDB to allow a multi-master MySQL cluster. I used a chef cookbook to deploy it[1]. [0] http://www.codership.com/content/using-galera-cluster [1] http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol -- Leif Madsen http://www.o

Re: [asterisk-users] PLAYIN MUSIC WHILE SEARCHING MYSQL

2012-09-28 Thread Leif Madsen
On 27/09/12 02:13 PM, Mehdi Rahimi wrote: On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen wrote: On 26/09/12 05:35 AM, Mehdi Rahimi wrote: I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Probably Local

Re: [asterisk-users] RealTime table fields ordering

2012-09-28 Thread Leif Madsen
ial source tarball. That of course also implies contributions to review the files prior to release (which have release candidates). That directory contains data that was at one point contributed, and should really be reviewed by the community with any changes required submitted back upstream.

Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

2012-09-28 Thread Leif Madsen
which he's using as an example in the upcoming Asterisk: The Definitive Guide 4e book. [0] https://github.com/russellb/amiutils -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

2012-09-28 Thread Leif Madsen
ps://issues.asterisk.org/jira/browse/ASTERISK-20150 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introdu

Re: [asterisk-users] PLAYIN MUSIC WHILE SEARCHING MYSQL

2012-09-26 Thread Leif Madsen
On 26/09/12 05:35 AM, Mehdi Rahimi wrote: I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Probably Local channels to the rescue here. -- Leif Madsen http://www.oreilly.com/catalog/asterisk

Re: [asterisk-users] confbridge video support

2012-09-25 Thread Leif Madsen
in ConfBridge(). Look at video_mode=follow_talker -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012

2012-08-27 Thread Leif Madsen
On 27/08/12 10:08 AM, Asterisk Development Team wrote: As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. Huzzuh! Does this mean http://issues.asterisk.org will now go directly to JIRA? Leif. -- Leif Madsen http

Re: [asterisk-users] SLA - Shared Line Appearance - Polycom

2012-05-15 Thread Leif Madsen
ns * usage of SLAtrunk() and SLAstation() applications to use the SLA lines, which also changes the device status (that is monitored by the device) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and

Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel

2012-04-27 Thread Leif Madsen
-AudioManipulation_id302347.html I also talked a bit about injecting audio onto a channel at AstriCon 2011 in my Cooking With Asterisk talk. It's the last recipe I talk about in this video: http://www.astricon.net/videos/Cooking-with-Asterisk.html -- Leif Madsen http://www.oreilly.com/catalog/ast

Re: [asterisk-users] Best non polycom SIP conference room phone

2012-01-09 Thread Leif Madsen
On 05/01/12 05:24 PM, Kevin P. Fleming wrote: Although in my personal opinion, it's really hard to beat the IP5000. That has been my experience as well. -- Leif Madsen http://www.oreilly.com/catalog/ast

Re: [asterisk-users] tcp version of toronto - osaka doesn't work

2012-01-09 Thread Leif Madsen
as expected. Also, please be sure to file errata so that we can look at it for the next printing or version of the book (depending on what the issue actually is). Errata can be filed at http://oreilly.com/catalog/9780596517342/errata/ -- Leif Madsen http://www.oreilly.com/catalog/asterisk

Re: [asterisk-users] PITCH_SHIFT()

2011-12-20 Thread Leif Madsen
ctions would be called dynamically from features.conf. You'd just create the application_map as documented in features.conf and then apply the PITCH_SHIFT() function to whichever channel you want. Untested, but should look something like: pitch_up_them => 3*,peer/

Re: [asterisk-users] Mixing asterisk.conf, asterisk.ael and asterisk realtime

2011-12-09 Thread Leif Madsen
iled" into Asterisk dialplan (like extensions.conf) anyways, so you can. You just have to follow the same rules about there not being duplicate macro names, as the AEL and dialplan logic is going to be combined together in memory. Leif. -- Leif Madsen http://www.oreilly.com/c

Re: [asterisk-users] Preparing to store vm in database

2011-12-09 Thread Leif Madsen
you're thinking, but it was never complete, and has never become a priority issue for any community developer to complete. You could of course create an AGI() script that pulled the audio out of the database, caching the audio for a period of time, then ju

Re: [asterisk-users] Frequent Asterisk Restarts

2011-11-10 Thread Leif Madsen
On 11-11-10 01:15 PM, Eric Wieling wrote: > The Asterisk source tree has a document with instructions on getting a > backtrace from the segfaults so you can report it on the issue tracker. Most up to date documentation is on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Debugging

Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

2011-11-10 Thread Leif Madsen
On 11-11-10 12:12 PM, Danny Nicholas wrote: > Yeah! My boss will be much happier having a system that doesn't have the > -tail on it. I hear this kind of statement every once in a while, which makes absolutely no sense to me. If you're blindly running a version of any software in production (rega

Re: [asterisk-users] Permanent sip and agi debug on?

2011-11-10 Thread Leif Madsen
On 11-11-09 04:30 PM, Danny Nicholas wrote: > If you have an "ancient" version of Asterisk you want to stick with, you can > do this with asterisk -rx "sip set debug on" and asterisk -rx "agi set debug > on" in your safe_asterisk script. Not sure about AGI, but pretty sure the sipdebug=yes option

Re: [asterisk-users] 1.8.8.0-rc2 Missing core functions.

2011-11-10 Thread Leif Madsen
On 11-11-07 08:38 AM, Bryant Zimmerman wrote: > I have a test box that has been running asterisk without issue. I updated > it to 1.8.8.0-rc2 and now I am getting some wierd issues I have never seen > before. > > > All the modules seem to have compiled without issue. > > > Asterisk starts up

Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

2011-11-10 Thread Leif Madsen
On 11-11-10 11:57 AM, Danny Nicholas wrote: > Misspoke - when should we expect 10.0 that is not -rc or -beta? Well Asterisk 10.0.0 is now in release candidate status, which means pending any major issues or regressions, a full Asterisk 10.0.0 is inevitably due in the near future. Leif. -- __

Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

2011-11-10 Thread Leif Madsen
On 11-11-10 11:43 AM, Danny Nicholas wrote: > Does this mean a "non-beta" labeled Asterisk 10.0 is due out shortly? I'm confused by shortly the announcement means it is out now. That was the purpose of the announcement... http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-10.0.0-rc

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

2011-09-20 Thread Leif Madsen
imply that the 's' extension was a catch-all. In the first and second editions of Asterisk: The Future of Telephony we were mostly using analog lines, and thus the usage of the extension 's' was fairly prominent. There are many other single letter extensions that have extra m

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

2011-09-20 Thread Leif Madsen
nge all of my simple dialplans to read exten=> start,1,blah instead of exten => s,1,blah. To me exten=> s,1,blah is more intuitive and less vulnerable than exten => _X.,1,blah. The 's' extension does stand for 'start' but I don't think we've eve

Re: [asterisk-users] Log for voicemail to email?

2011-09-20 Thread Leif Madsen
ble you haven't installed an MTA or have it disabled. Or perhaps the other ends are rejecting due a missing MX record, or some other email configuration issue.) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- __

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

2011-09-13 Thread Leif Madsen
2/asterisk-Install.html#Installing_id291070 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar e

Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8

2011-09-13 Thread Leif Madsen
you define "NAT problem"? I'm unaware of any issues with Asterisk (or end points) behind NAT. It is mostly likely a configuration issue rather than a bug. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _

Re: [asterisk-users] Asterisk on Android?

2011-09-08 Thread Leif Madsen
On 08/09/11 02:19 PM, Cobra 2 wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. I think the question is, can you answer your incoming calls with the Asterisk running on the device? -- Leif Madsen http://www.o

Re: [asterisk-users] trying to build 1.8.6.0 on CentOS 6, problems with ptlib

2011-09-06 Thread Leif Madsen
along as I thought it was pretty neat when I learned about it :)) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] Set(CHANNEL(musicclass)=

2011-09-06 Thread Leif Madsen
,NoOp() same => n,Set(CHANNEL(musicclass)=default) I could use just: exten => s,n,MusicOnHold() There is a lot of documentation on www.voip-info.org but sometimes it is not clear which asterisk version it applies to :-/ (Another good reason to be reading the documentation on https://wiki.aster

Re: [asterisk-users] Beggining asterisk

2011-09-06 Thread Leif Madsen
On 04/09/11 02:51 PM, Tamer Higazi wrote: the 3rd edition is available, but that book covers every thing to run the asterisk PBX. You can read the 3rd edition online at http://ofps.oreilly.com/titles/9780596517342/ HTH! Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk

Re: [asterisk-users] dialplan pattern help

2011-07-23 Thread Leif Madsen
) same => n,ExecIf($["${EXTEN:-8}" = "42704701"]?Macro(dialfax,${EXTEN:-8})) same => n,Verbose(2,Did not match -- falling through) same => n,Playback(invalid) same => n,Hangup() I'm pretty sure that's the only way you can do it in a single line (th

Re: [asterisk-users] Next Asterisk 1.8 Release

2011-06-19 Thread Leif Madsen
sting changes in your development systems at any period of time. 1.8.5 release candidates should be available later this week though. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Coloc

Re: [asterisk-users] MixMonitor

2011-06-16 Thread Leif Madsen
K, well nothing looks obviously wrong there from what I can tell. What is your console output doing though when you do the transfer? Are you using Asterisk transfers? What version of Asterisk are you using? Leif. -- Leif Madsen http://www.oreilly

Re: [asterisk-users] MixMonitor

2011-06-16 Thread Leif Madsen
came in from the ITSP. If that channel is then transferred, the recording should follow it around. Can you elaborate a bit more on the call flow and show the console output? -- Leif Madsen http://www.oreilly.com/cata

Re: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)

2011-06-02 Thread Leif Madsen
On 02/06/11 03:35 PM, satish patel wrote: Is this available in current SVN ? Changes are always checked into SVN first and then made available in a tag. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk

Re: [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue

2011-05-27 Thread Leif Madsen
log/9780596517342 (left hand side). That way we can get it fixed up in subversion. Thanks! Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Asterisk 1..8 multiple queue

2011-05-26 Thread Leif Madsen
-ACD.html#ACD_id288626 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] MagicJack quality

2011-05-24 Thread Leif Madsen
On 11-05-24 01:21 PM, Steve Edwards wrote: > If it take the OP (of this thread) 3 years to reply, what does that say about > their product support? Par for the course? :) Leif. -- _ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

2011-05-20 Thread Leif Madsen
On 11-05-20 10:39 AM, Benoit Panizzon wrote: > After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is > just put in a temporary variable __SIPDIVERSIONREASON but not in a variable > useable in the dialplan. You could double check by using DumpChan() to see what channel var

Re: [asterisk-users] AstManProxy

2011-05-20 Thread Leif Madsen
On 11-05-20 09:37 AM, Ishfaq Malik wrote: > Do many people use this? > Is it reliable and safe? It may still work, but that code is quite old, and I'm not even sure it's necessary any more. Leif. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Different box for SIP and RTP

2011-05-16 Thread Leif Madsen
On 11-05-16 09:13 AM, Alex Balashov wrote: > On 05/16/2011 09:00 AM, Mohammad Khan wrote: > >> Is there way I can use two Asterisk box, one to maintain SIP packets and >> other for RTP traffic? > > No, the signaling and bearer plane are integrated in Asterisk. > > But you can use reinvites to ha

Re: [asterisk-users] Backport of DEVICE_STATE to 1.4

2011-05-16 Thread Leif Madsen
On 11-05-16 07:29 AM, Olivier wrote: > As this bug is considered "fixed", I think you can't add any comment > anymore. > Unfortunately, you can still see lines mentionning DEVSTATE function like : > > if (ast_strlen_zero(data)) { > ast_log(LOG_WARNING, "DEVSTATE function called

Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so

2011-05-15 Thread Leif Madsen
On 11-05-13 11:39 AM, isr...@gmail.com wrote: > I haven't tried with timerfd but with timer pthread 1.8 is very unstable > > I think I have seen a post to the list from kevin fleming that the same is > for timerfd that there is a nasty bug which they haven't found the reason for > yet My exper

Re: [asterisk-users] Asterisk 1.8.4 Now Available

2011-05-12 Thread Leif Madsen
On 11-05-11 09:31 PM, Jose P. Espinal wrote: > Download links on the website have not been updated (asterisk.org) Oops sorry! I will fix that right.. now! Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] concurrent call tracking

2011-05-12 Thread Leif Madsen
On 11-05-11 06:36 PM, Skyler wrote: > Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd > like to take a look at it for sure. The dial plan example Leif replied with > is pretty much what I was thinking, just didn't have a clue how to go about > it. ;) You could also look

Re: [asterisk-users] CLI - displaying all channel variables

2011-05-11 Thread Leif Madsen
On 11-05-11 12:29 PM, Steve Edwards wrote: > On Wed, 11 May 2011, Eric Wieling wrote: > >> Generally you should insert a Noop in the dialplan to examine variables. >> Noop(EXTEN is ${EXTEN}) for example. > > The 'verbose()' application would be an example of 'better practices.' > > It's function

Re: [asterisk-users] concurrent call tracking

2011-05-11 Thread Leif Madsen
On 11-05-11 12:57 PM, Skyler wrote: > I would like to track/store concurrent call usage per user by > day/week/month and get server totals by day/week/month. Google comes up with > mostly info regarding concurrent call limits, though my goal is to calculate > actual concurrent channel usage and ad

Re: [asterisk-users] Asterisk 1.8.4 Now Available

2011-05-11 Thread Leif Madsen
this case, tags/1.8.4-rc3 as copied to tags/1.8.4, and the only changes were made to the .version file and ChangeLog. Then the standard release process is followed to turn that tag into a .tar.gz and get it onto the downloads site. Any changes made

Re: [asterisk-users] question on ways to activate voicemail light on polycom

2011-05-07 Thread Leif Madsen
On 11-05-06 02:56 PM, Watkins, Bradley wrote: > Yes, use the MinivmMWI application. That's how I've done it in the past as well. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? J

Re: [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend

2011-05-03 Thread Leif Madsen
On 11-04-30 03:10 PM, Alec Taylor wrote: > Good Evening, > > I'm setting up an Internet Radio website with call-in functionality, > and need to know the kinds of FOSS tools I should install to get the > job done. > > Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png >

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Leif Madsen
On 11-04-29 02:59 AM, Olle E. Johansson wrote: > > 29 apr 2011 kl. 01.49 skrev Leif Madsen: > >> Well the issue is that we currently have over 900 open issues in the Asterisk >> project alone, and with only one primary bug marshal (myself) sometimes >> things >&

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Leif Madsen
On 11-04-28 07:09 PM, Alec Davis wrote: > Making an assumption here, I'm sure I cleared the remaining resequencing > issues up in 1.4 SVN and 1.6.2 SVN. > https://issues.asterisk.org/view.php?id=19032 > > The issues I uncovered and fixed were when a new voicemail is left, while a > mailbox is open

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Leif Madsen
On 11-04-28 07:02 PM, Ira wrote: > At 03:48 PM 4/28/2011, you wrote: > OK, maybe not, but if I thought it was a bug and you discover it was a bug and > fix it, than who was it who decided it wasn't a bug 15 minutes after I put it > in > the bug tracker and why did that person have that much power?

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Leif Madsen
On 11-04-28 04:33 PM, Administrator TOOTAI wrote: > Le 28/04/2011 21:47, Leif Madsen a écrit : >> On 11-04-28 12:04 PM, Administrator TOOTAI wrote: >>> Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes >>> for >>> few weeks/monthes

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-28 Thread Leif Madsen
On 11-04-28 12:04 PM, Administrator TOOTAI wrote: > Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for > few weeks/monthes till 1.8 reaches the level that the community accept to > switch > to 1.8 What is the guide here? What is the "level that the community" accepts? U

Re: [asterisk-users] core show channels consise in asterisk 1.8.3

2011-04-18 Thread Leif Madsen
On 11-04-18 02:47 PM, Jerry Geis wrote: > When I do "core show channels concise" over the AMI interface > how do I specify that I want to see the "actual" channel number like > DAHDI/4/xxx > where 4 is the actual channel. > > RIght now I am seeing DAHDI/i1/x where i1 is the span. I co

Re: [asterisk-users] Asterisk 1.8.3

2011-04-07 Thread Leif Madsen
On 11-04-07 09:45 AM, --[ UxBoD ]-- wrote: And don't forget that call pickup crashes Asterisk from what would appear release 1.8.1 upwards! We have had to back level to that latest 1.6 branch. https://issues.asterisk.org/view.php?id=18654 I ran into this issue as well on 1.8.3.2, but I didn't

Re: [asterisk-users] SIP Provider Recommendation in US

2011-03-03 Thread Leif Madsen
On 11-03-03 11:22 AM, Brent A. Torrenga wrote: I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-ni

Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8

2011-03-01 Thread Leif Madsen
On 11-02-27 09:12 PM, Stuart Longland wrote: I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here.

Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

2011-02-25 Thread Leif Madsen
On 11-02-24 08:56 PM, Andrew Latham wrote: And I go back to triple check and compare revision numbers... You are 100% correct, the revision numbers in our local repository are wrong, someone pushed the 1.8.3 RC3 into our local 1.8.2 branch. I apologize and will work to better control my trust o

Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

2011-02-24 Thread Leif Madsen
On 11-02-24 04:08 PM, Andrew Latham wrote: There are many updates in 1.8.2.4 that may fix your issue. If you are running any version of 1.8 it should be a quick update. I wouldn't say "many". There is one fix in 1.8.2.4 over 1.8.2.3. From the ChangeLog: * Asterisk 1.8.2.4 Released.

Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Leif Madsen
On 11-02-23 10:31 AM, Jose P. Espinal wrote: Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Actually I was wrong! See here. It is being resolved. https://reviewboard.asterisk.org/

Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Leif Madsen
On 11-02-23 10:31 AM, Jose P. Espinal wrote: - Added a new configuration option "remotesecret" for authentication to remote services. For backwards compatibility, "secret" still has the same function as before, but now you can configure both a remote secret and a local secret for mutual authe

Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

2011-02-22 Thread Leif Madsen
On 11-02-22 10:16 AM, Ishfaq Malik wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 No. The ChangeLog would give you the information you're looking for. http://downloads.asterisk.org/

Re: [asterisk-users] Hide the plain text password

2011-02-14 Thread Leif Madsen
On 11-02-14 05:08 PM, Jian Gao wrote: I am building a server for a client. I want them to try out the new Google Voice feature using my GV account. But I don't want expose my GV's password. Actually in this case, your best bet is just going to be to create a separate account where you don't ca

Re: [asterisk-users] Hide the plain text password

2011-02-14 Thread Leif Madsen
On 11-02-14 05:10 PM, Kevin P. Fleming wrote: On 02/14/2011 04:08 PM, Jian Gao wrote: I am building a server for a client. I want them to try out the new Google Voice feature using my GV account. But I don't want expose my GV's password. There is no method to obscure a Google Voice password in

Re: [asterisk-users] [modules.conf] Modules still loaded after "noload"

2011-02-13 Thread Leif Madsen
On 11-02-13 09:52 AM, Gilles wrote: I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: Does someone know why Asterisk still loads modules even with the above lines in modules.conf? It looks like you're loading Asterisk, which loads all the modules, then modifyin

Re: [asterisk-users] Asterisk Performance

2011-02-01 Thread Leif Madsen
On 11-02-01 05:22 PM, Juan David Diaz wrote: I would like to handle about 250 simultaneous (calls& agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cac

Re: [asterisk-users] Asterisk 1.8.2.3 Now Available

2011-01-26 Thread Leif Madsen
On 11-01-26 04:07 PM, Kevin P. Fleming wrote: On 01/26/2011 03:06 PM, Warren Selby wrote: Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new versioning methods made updates into 1.8.x releases and security updates into 1.8.x.y releases? Security fixes and regression fi

Re: [asterisk-users] Recommended Windows client to display CID?

2011-01-26 Thread Leif Madsen
On 11-01-26 08:52 AM, Gilles wrote: Hello I'd like to display CID information on users' monitor running Windows. You could use any XMPP client and send a message to it using JabberSend() from the dialplan. We document using it at http://ofps.oreilly.com. Leif. -- __

Re: [asterisk-users] Asterisk on Debian Lenny with timerfd

2011-01-24 Thread Leif Madsen
On 11-01-24 08:29 AM, RR wrote: Wow, alright, after an all-nighter, I was able to get timerfd.so compiled in Asterisk 1.8.2.2 under Debian Lenny 5.0.7 with Kernel 2.6.26-2-amd64. Of course, due to the glibc requirement of 2.8+, a lot of dodgey upgrades had to be performed. I have no idea how "sta

Re: [asterisk-users] Info on using LDAP with Asterisk?

2011-01-24 Thread Leif Madsen
On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517

Re: [asterisk-users] MOH and parking

2011-01-21 Thread Leif Madsen
On 11-01-21 08:52 AM, Andrew Thomas wrote: I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). After speaking with Shaun and Russell, this is lik

Re: [asterisk-users] HA: what is missing to keep ongoing calls during failover ?

2010-12-17 Thread Leif Madsen
On 10-12-17 06:17 AM, Olivier wrote: Hi, What is currently missing in Asterisk ecosystem to get 2 servers active-active redundancy such as when server 1 is failing (in some circumstances), its ongoing calls (or most of them) are kept alive and handed over to server 2 ? I remember that a couple

Re: [asterisk-users] Call sip:u...@domain.com?

2010-12-17 Thread Leif Madsen
On 10-12-17 06:48 AM, Gilles wrote: On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI wrote: Then create a prefix for SIP calls exten=>_9.,1,Dial(SIP/${EXTEN:1}) and you dial 9u...@domain.com from XLite Remember that calling sip URL is not as easy with a phone. Imagine you have an A

Re: [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8

2010-12-16 Thread Leif Madsen
On 10-12-15 09:46 AM, bilal ghayyad wrote: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? It depends on your required usage (features available in version) and your required s

Re: [asterisk-users] wideband recording in Asterisk 1.8

2010-11-23 Thread Leif Madsen
On 10-11-23 08:24 AM, Henry Dogger wrote: > I have an aastra 6739i which supports the g722 codec. > > Which format setting do I need to be able to record in wideband? > > Tried: wav, gsm, pcm. Nothing seems to give me the result I desire. Shouldn't you try g722 as the format? Leif. -- _

Re: [asterisk-users] Asterisk 1.8 Release Schedule

2010-11-23 Thread Leif Madsen
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote: > I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta > or release candidate ? It will be Asterisk 1.8.1-rc1 and that is now available (as of a few minutes ago) http://www.asterisk.org/node/51466 Leif. -- __

Re: [asterisk-users] Best way to connect to a MySQL Database

2010-11-15 Thread Leif Madsen
On 10-11-15 06:04 PM, Matt Darnell wrote: > Is this command the best way to access a MySQL database - > MYSQL(Connect connid dhhost dbuser dbpass dbname) ? > > I thought I heard that using ODBC was a bit more stable. > > Anyone have any experience? Use func_odbc along with res_odbc. I've taken dia

Re: [asterisk-users] Volume on meetme recording

2010-11-15 Thread Leif Madsen
On 10-11-15 08:30 AM, Richard Kenner wrote: > It's kind of low for me. How does one control that volume? You could use the VOLUME() function prior to joining the conference for channels that are quiet. Leif. -- _ -- Bandwidt

Re: [asterisk-users] Extension Exists

2010-10-25 Thread Leif Madsen
On 10-10-25 04:21 PM, Dan Journo wrote: > Hi, > > When a VOIP user dials an external number, the calls are routed through > our SIP provider. > > Is there a simple way to check whether the DDI exists locally before > dialling out to the sip provider? > > Something like GotoIfExists(5551...@incoming

Re: [asterisk-users] dahdi vmware query

2010-10-19 Thread Leif Madsen
On 10-10-19 10:46 AM, Danny Nicholas wrote: > Greeting list, > > I hope this isn’t a “lazy” question. I have been running TDM400P and > TDM410P cards in Dell PowerEdge Servers for a few years now. We are > moving from physical servers to VMWARE servers. What opportunities > should I expect moving t

Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available

2010-10-19 Thread Leif Madsen
On 10-10-18 11:01 PM, Barry Miller wrote: > On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote: >> On 10-10-18 07:54 PM, Asterisk Development Team wrote: >>> For a full list of changes in the current release candidate, please see the >>> ChangeLog: >>> >>> http://downloads.as

Re: [asterisk-users] fraud advice

2010-10-15 Thread Leif Madsen
On 10-10-15 07:20 AM, Leif Madsen wrote: > On 10-10-14 10:49 PM, bruce bruce wrote: >> Unfortunately, probably there is no one you can complain to. But it also >> sickens me at how badly Asterisk is made to not cope with situations >> like this and worse than that is FreePBX

Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone

2010-10-15 Thread Leif Madsen
On 10-10-15 04:10 AM, Сикорский Сергей wrote: > 15.10.2010 9:40, Warren Selby пишет: >> I think this means you need to set a call-limit for each sip peer > > Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter

Re: [asterisk-users] fraud advice

2010-10-15 Thread Leif Madsen
On 10-10-14 10:49 PM, bruce bruce wrote: > Unfortunately, probably there is no one you can complain to. But it also > sickens me at how badly Asterisk is made to not cope with situations > like this and worse than that is FreePBX. How is password policy an Asterisk issue? The solution to the probl

Re: [asterisk-users] Routers that do not show external IPs...

2010-10-14 Thread Leif Madsen
On 10-10-14 12:18 PM, Carlos Chavez wrote: > I have a customer that has a Trendnet TEW-435BRM router which has the > bad habit of rewriting all external connections so the Asterisk server > only sees the IP address of the router itself. Up to today this has not > been a problem since all ext

Re: [asterisk-users] Module reload

2010-10-04 Thread Leif Madsen
On 10-10-04 10:59 AM, Flavio Miranda wrote: > > > Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf > > ; DAHDI telephony > ;language=en > ;echocancel=yes > echocancelwhenbridged=yes > > ss7type = itu > ss7_called_nai=dynamic > ss7_calling_nai=dynamic > > ;General options > usecallerid

Re: [asterisk-users] Module reload

2010-10-04 Thread Leif Madsen
On 10-10-04 09:44 AM, Flavio Miranda wrote: > Hi all, > > Every time I reload my asterisk it fall down and the following message > appear on log: > > parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf > > If I comment that line, it change to other line. > > There are some

Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available

2010-09-27 Thread Leif Madsen
On 10-09-26 02:55 PM, Ira wrote: > At 10:37 PM 9/24/2010, you wrote: >> You probably need to install libssl-dev then rerun ./configure. At >> least I did (Debian Lenny). Seems chan_sip needs res_crypto which >> needs libssl. > > Thanks, I tried to figure out what I needed but I failed. That was >

Re: [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality

2010-09-27 Thread Leif Madsen
On 10-09-26 01:00 PM, bilal ghayyad wrote: > First of all, I am looking to have the H323 Gatekeeper service available at > Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing > H323 gatekeeper functionality or not? > > Until 1.4.26.2 version, there is no h323 gatekeeper f

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