On Monday 17 November 2008 10:38:30 am Julian Lyndon-Smith wrote:
> Is it possible to use hints from a realtime source like a db or curl ?
> I was looking at the grandstream GXP2000 Expansion Module (EXT) which
> has 56 fully programmable keys that work with BLF. You can daisy-chain 2
> of these to
On Monday 17 November 2008 03:57:36 am Veselin K wrote:
> Could you please tell me what is the process called so I can
> research it further.
The field you're looking for in the database is called "fullcontact". Also,
note that if you don't have this field or if the field is too short (should be
On Monday 17 November 2008 02:26:21 am Mikel Lindsaar wrote:
> 2) Also wondering what people do when parsing asterisk verbose output in
> the log. Specifically, following a certain call. Asterisk's verbose
> output logs in sequence of action, which is good, but if you have 40-50
> workstations go
For a long while, Asterisk administrators have had the desire to take PRI
channels out of service and keep them that way, usually to create a window
in which to perform maintenance or another such purpose. We are finally very
close to having that functionality in Asterisk, but what we need is a fe
On Friday 14 November 2008 09:19:22 Gordon Henderson wrote:
> On Fri, 14 Nov 2008, Tzafrir Cohen wrote:
> > On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote:
> >> I used to use IDEFISK, but since it was taken over/renamed into Zoiper
> >> it's been really hard work - now I'm told th
On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
> Is there somewhere a statement from Digium how long they will support
> Asterisk 1.4?
There is no statement, because we haven't even discussed when the EOL for
1.4 will be reached. Certainly that means it won't happen for at least the
On Wednesday 12 November 2008 23:44:49 Eric "ManxPower" Wieling wrote:
> Tilghman Lesher wrote:
> > On Wednesday 12 November 2008 23:15:01 Eric "ManxPower" Wieling wrote:
> >> Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
> >>
On Wednesday 12 November 2008 23:15:01 Eric "ManxPower" Wieling wrote:
> Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
> chan_iax (maybe other channels) translate the protocol specific causes
> to a Q.931 hangup cause.
>
> I would recommend you check the doc directory in the
On Wednesday 12 November 2008 18:34:45 Steve Totaro wrote:
> On Wed, Nov 12, 2008 at 6:57 PM, Alex Balashov
<[EMAIL PROTECTED]>wrote:
> > Steve Totaro wrote:
> > > I believe that if you are speaking of code and Asterisk's
> > > implementation of the SIP RFC it is already very borked in many many
>
On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
> Jared Smith wrote:
> > On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
> >> Anyone looking for firmware should get it now before it disappears.
> >
> > It's my understanding that this isn't a fluke, but that Polycom has
> > indee
On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
> On 11/11/2008 10:48 p.m., samuel wrote:
> > So far I've updated a few machines (1.4.22) and the DNS queries are
> > reduced to a minimum, at least haven't seen DNS channel queries...
>
> Slightly more useful to me, does anyone know the patc
On Monday 10 November 2008 05:56:18 Tony Mountifield wrote:
> Does anyone here know anything about GEN-GEN analogue circuits, also
> known as Manual Ring-Down (MRD)? Apparently they are widely used in
> Hoot'n'Holler systems for financial dealer-boards.
>
> I have been asked to try and interface to
On Thursday 06 November 2008 23:45:38 vivek rastogi wrote:
> I need a help on g729 codec.Is there any tool which can convert g711 codec
> into g729 codec and supports batch processing ?
for from in /full/path/to/directory/*.ul ; do
to=${from%%.ul}.g729
asterisk -rx "file convert $f
On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
> Dou you have any example? Can I call directly to querys without the
> templates???
func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something
extensions.conf:
exten => 123,1,Set(result=${ODBC_EXEC(SELECT foo FROM bar)})
--
On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote:
> The main thing is that with this I set the Store Procedure or Query
> directly on the dialplan line, is easier to configure, change, manage, etc.
>
> I also know that works great with heavy load, and it reconnects when the
> network
On Thursday 06 November 2008 11:41:08 Brent Davidson wrote:
> Tilghman Lesher wrote:
> > [companyA]
> > exten => _X.,1,Set(company=A)
> > exten => _X.,n,Goto(maincontext,${EXTEN},1)
> >
> > [companyB]
> > exten => _X.,1,Set(company=B)
> > ex
On Thursday 06 November 2008 10:12:11 Brent Davidson wrote:
> If I have a global variable in my dialplan and I change it, does that
> change immediately take affect for all calls that are active?
>
> Here is my situation. The company I work for has two office groups that
> share a building. The t
On Thursday 06 November 2008 08:53:40 Louis-David Mitterrand wrote:
> On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
> > Louis-David Mitterrand wrote:
> > > When monitoring an asterisk through its iax2 port I get these warnings
> > > at the console:
> > >
> > > [Nov 6 13:15:15
On Tuesday 04 November 2008 10:30:10 Jeremy Mann wrote:
> Is there a way to override sip peers defined in users.conf with respect to
> their context and hints?
No, there is not. Users.conf is meant to be a very simple interface for
adding users, designed especially for the Asterisk GUI project.
On Tuesday 04 November 2008 16:02:40 Jeff LaCoursiere wrote:
> On Tue, 4 Nov 2008, Dima wrote:
> > The person I'm giving the access to is an admin of that asterisk. It's
> > up to him to do evil stuff with asterisk itself. as long as he can't get
> > a shell and do "rm -rf /" I'm safe.
>
> Hmm, I w
On Tuesday 04 November 2008 15:52:10 Ruddy Gbaguidi wrote:
> Did you know that any commandyou type in asterisk cli starting with
> exclamation point (!) is execute in the shell by asterisk ??
> Example :
> running
> !ls
> will run 'ls' in your current directory
>
> So, be aware because your user ca
On Tuesday 04 November 2008 12:37:03 Leah Newmark wrote:
> Thanks!
> Apparently, I forgot to mention our version running is 1.2.13
> Is the coding similar enough to give it a shot?
No, the code is likely to be completely different. In fact, the trouble
you're having may just be the reason why it
On Monday 03 November 2008 12:19:31 Leah Newmark wrote:
> The directory is owned by asterisk and permissions seem fine there too:
> drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 .
>
> I never see a voicemail.conf.new created. I have done a locate on the
> whole server and don't see a blank o
On Saturday 01 November 2008 18:52:41 Alexander Lopez wrote:
> No need to compile "!" out of asterisk source
>
> Just put SHELL=/bin/false in your login script
>
> The ! command will not work...
That's not completely true. The only thing that will prevent is the ability
to get a shell pro
On Wednesday 29 October 2008 19:10:49 joe mcguckin wrote:
> I recently built Asterisk from scratch on Ubuntu (Ubuntu
> 4.2.3-2ubuntu7). Everything seemed to build ok, but when I start
> Asterisk, I get the message:
>
> Warning! Asterisk is not thread safe.
>
> Is this anything to be concerned
On Wednesday 29 October 2008 10:22:43 David Gibbons wrote:
> A phone takes very, very little bandwidth away from the desktop and a decent
> one will support tagging its frames for the alternate voice VLAN.
>
> --snip--
> In almost all cases it is much better to have two seperate networks.
> This ma
On Sunday 26 October 2008 21:28:34 Andrew Joakimsen wrote:
> Other vendors, including Cisco, will provide the firmware directly. I
> no longer deploy Polycom (unless someone really wants them) due to
> this. Yes I can get it from the supplier but it takes a few days. I
> would rather just go to Pol
On Wednesday 22 October 2008 12:27:17 Nhadie wrote:
> hs anyone able to make video to work on asterisk? i tried following this:
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
>
> i can see that eyebeam is trying to broadcast a video but the other
> eyebeam is not recei
On Tuesday 21 October 2008 13:33:42 Armand Fumal wrote:
> The problem is in asterisk -r
There is no such problem. All 'asterisk -r' does is to take output from
a pipe and sends it to your terminal. It is the core Asterisk process which
is failing to generate terminal color codes.
> My terminal
On Tuesday 21 October 2008 11:43:31 Armand Fumal wrote:
> Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in
> console. Do I miss a package or compilation option ?
Neither. The startup scripts changed slightly, such that they no longer
output to a terminal. Because there is
On Monday 20 October 2008 20:42:30 jordan pan wrote:
> I have encountered a hard problem that when i connect my anology phone
> to channelbank ,I found that i dial a number and create the call,then ,I
> hangup the call,but ,very quickly,I listen the ringing im my phone,I pick
> it up ,and found
On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote:
> I need help in implementing Asterisk with LDAP. I' ve installed Asterik
> 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra
> LPAD.
You might want to take a look at Asterisk 1.6, which has LDAP realtime
support. Look w
On Friday 17 October 2008 11:46:09 Brian J. Murrell wrote:
> On Fri, 2008-10-17 at 11:35 -0500, Eric "ManxPower" Wieling wrote:
> > If you fight Asterisk's oddities then you will have a depressing and
> > miserable life. If you embrace Asterisk's oddities then you will have a
> > joyous and enligh
On Friday 17 October 2008 10:15:22 Olivier wrote:
> 2008/10/17 John Todd <[EMAIL PROTECTED]>
>
> > At 5:46 PM +0200 2008/10/16, Olivier wrote:
> > >Is Incomplete() application an acceptable work around for ISN ?
> >
> > It is impossible to determine the full sequence of digits for an ISN
> > number
On Friday 17 October 2008 10:32:30 michel freiha wrote:
> I tried to put + before the x like _+X but when making a call i got the
> following error:
>
> <>
> [Oct 17 15:08:58] WARNING[17532]: ast_expr2.fl:407 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected '+', exp
On Friday 17 October 2008 01:09:18 Lee, John (Sydney) wrote:
> So, at this stage, my res_config_mysql.c is still not writing anything
> into table queue_log despite having: a) correct res_mysql.conf b)
> extconfig.conf c) mysql up and running d) res_config_mysql.c start up
> okay
>
> I believe that
On Thursday 16 October 2008 13:59:46 Karl Fife wrote:
> On Thu, 16 Oct 2008 11:47:15 -0500, "Tilghman Lesher"
>
> <[EMAIL PROTECTED]> said:
> > If you could explain what ISN is, that might help.
>
> an ISN, stands for ITAD Subscriber Number, which in
On Thursday 16 October 2008 13:38:00 Ken D'Ambrosio wrote:
> Hi, all. This e-mail is a follow-up to an exchange I had several weeks
> ago. I've got an Asterisk box with a dual-span T1 card. I want to place
> it between the PSTN and my company's legacy PBX. I actually did do that,
> but internat
On Thursday 16 October 2008 10:46:51 Olivier wrote:
> Is Incomplete() application an acceptable work around for ISN ?
If you could explain what ISN is, that might help.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Wednesday 15 October 2008 15:59:09 jonathan augenstine wrote:
> Is it possible to create extensions in the voicemail.conf remotely by using
> the manager interface. I cannot seem to find any documents or examples
> describing that capability.
It's not possible, no. However, you could enable r
On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote:
> For some calls (usally telemarketers) entering through a BRI zap channel
> I somtimes notice the callerid on my polycom 601 phone and the CDR's
> 'src' field don't match. They are even totally different. And the
> displayed calle
On Tuesday 14 October 2008 02:10:39 Olivier wrote:
> 2008/10/13 Tilghman Lesher <[EMAIL PROTECTED]>
>
> >
> >
> > Pray tell, how do you echo cancel in both directions? Wouldn't that
> > necessitate cancelling echo before it occurs on the line (sor
On Tuesday 14 October 2008 05:00:45 Chris Rowson wrote:
> > I'm working on a solution using the Asterisk voicemail component and
> > wondered if anyone knew the answer to this question please?
> >
> > I understand that Asterisk saves voicemail to
> > /var/spool/asterisk/voicemail///INBOX/ but I
>
On Tuesday 14 October 2008 08:29:18 Rodolfo Alcazar Portillo wrote:
> 1xx -- office extensions
> 92xx -- local city calls, which must be free
> 90800xx. -- national 0800 calls, which must be free
> 4___#9xxx. other calls, which must be paid: cell, national,
> international. the three "_
On Monday 13 October 2008 18:57:41 Lee, John (Sydney) wrote:
> By the way, did you see anything wrong with my config files?
>
> /etc/asterisk/res_mysql.conf
> [general]
> dbhost = localhost
> dbname = db1
> dbuser = user
> dbpass = password
> dbport = 3306
> dbsock = /var/lib/mysql/mysql.sock
You
On Sunday 12 October 2008 04:15:02 Olivier wrote:
> 2008/10/11 Eric ManxPower Wieling <[EMAIL PROTECTED]>
>
> > Handsets use a 4-wire connection. Handsets with the the volume turned
> > up could cause a form of echo as the microphone picks up the ear piece
> > audio (I call this acoustic echo). E
On Monday 13 October 2008 11:19:35 gary wrote:
> - Original Message -
> From: "Tilghman Lesher" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, October 13, 2008 11:54 AM
> Subject: Re: [ast
On Monday 13 October 2008 10:29:17 gary wrote:
> I am running asterisk 1.2.27 and it dead today. The following is the
> backtrace of core file. Can anybody help me to identify what is the
> possible cause of crash? It seems the mysql connection causing problem in
> Thread 2. But I can not tell what
On Friday 10 October 2008 15:42:34 gary wrote:
> I am running asterisk 1.2.27 and it dead today. The following is the
> backtrace of core file. Can anybody help me to identify what is the
> possible cause of crash? It seems the mysql connection causing problem in
> Thread 2. But I can not tell what
On Wednesday 08 October 2008 22:05:14 Rob Hillis wrote:
> Tilghman Lesher wrote:
> > On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
> >> Wesley Haut wrote:
> >>> Yell at me if you will, but I hate func_realtime - it's not very
> >>> usable
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
> Wesley Haut wrote:
> > Yell at me if you will, but I hate func_realtime - it's not very
> > usable nor is it change-friendly (update your database and your
> > dialplan completely breaks).
>
> I agree completely. As it stands, the REALTIME(
On Tuesday 07 October 2008 21:58:31 Karl Fife wrote:
> So that leaves only one question:
>
> exten => ?
>
> What extension the following:
> '3129842314'
> '*989'
> '+13129842314'
>
> BUT does not match:
> 'i'
> 'james'
>
> is this possible?
I think you already described it in your original post:
On Wednesday 08 October 2008 02:20:38 Rob Hillis wrote:
> Tilghman Lesher wrote:
> >> Can someone suggest the best way to deal with this without resoring to a
> >> highly repetitive/iterative dialplan?
> >
> > Leif and I discussed something like this at Astricon 2
On Monday 06 October 2008 14:58:09 Karl Fife wrote:
> In several places online, and in the Asterisk F.O.T. book, there is a
> warning against using '_.' saying:
> "[it] should probably never be used".
>
> However, the need often arises act on numeric extensions that begin with
> *'s and #'s, and '+
On Saturday 04 October 2008 11:47:01 Guillermo V. Salas wrote:
> How can I prevent a remote DoS as described on the following site? :
>
> http://www.voip0day.com/news/remote-denial-of-service-exploit-effects-the-a
>sterisk-pbx/
This has already been addressed in the following advisory:
http://down
On Wednesday 01 October 2008 23:58:41 Max Alex wrote:
> Hi All,
> i have centos5 system, i have installed asterisk 1.4 branch.
> i havedone realtime connection with odbc to pgsql.
> i have created custom functions in func_odbc.conf, all dsn setup and
> connection is working fine,
> but custom funct
On Tuesday 30 September 2008 22:34:30 Andrew Joakimsen wrote:
> On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
>
> <[EMAIL PROTECTED]> wrote:
> > It is completely illegal in any country that recognizes patents.
>
> You mean countries that recognize software patents, r
On Wednesday 24 September 2008 19:28:17 Michael J. Liberatore wrote:
> Hi, i am running a small personal asterisk server for my business, and
> instead of getting a dedicated machine to run linux which would waste
> power and money i decided to run it on my windows xp sp2 machine. The
> machine is
On Saturday 20 September 2008 17:21:16 George Williams wrote:
> I have inherited some code that appears to implement a kluge-y way of
> adding and removing extensions, sip devices, and sip registrations
> dynamically.
>
> Yep, you guessed it - it modifies the extensions.conf and sip.conf files,
> a
On Thursday 18 September 2008 20:56:58 Craig Guy wrote:
> I felt at the time the written portion was heavily biased towards people
> who had done the training - in fact I would go so far as to say that it was
> designed specifically to discriminate against people who had not attended
> the official
On Friday 19 September 2008 14:54:58 George Williams wrote:
> I am issuing CLI commands via script, using the "asterisk -rx" method.
>
> Its working great. Now, I need to get the results of the command to look
> for error messages, etc.
>
> I've tried setting several "-v" flags as well, but I only
On Thursday 18 September 2008 18:13:44 Jay R. Ashworth wrote:
> How those are handled if you record your own arrangement of Hey Jude once
> and loop it on music on hold, I'm not clear on.
It's exactly the same, whether it's your own recording or not, since you did
not compose the music or the lyri
On Thursday 18 September 2008 14:19:29 David A. Bandel wrote:
> System language is spanish (es) and when users check their voicemail,
> if they don't delete it it goes into the Old directory.
>
> That's all well and good, but those users with messages in their Old
> directory try to get into voicem
On Thursday 18 September 2008 09:54:39 Steve Anness wrote:
> So here is the deal. I have an Asterisk server here at work that I
> have recently taken over and the boss is wanting the server to do a
> lot of things that it didn't do before. I have already configured much
> of what he wanted includin
On Thursday 18 September 2008 05:16:21 Olivier wrote:
> Whatever the verbosity level (even 0), my Asterisk console is full of
> "Really destroying SIP dialog" messages.
> Is there a way to get rid of those ?
Turn off debugging: core set debug 0 (and don't specify -d on your command
line).
> If n
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
> On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno <[EMAIL PROTECTED]> wrote:
> > Anybody knows how to get a Coupon Code for the discount on the Asterisk
> > training classes??? I am interested on taking that upcoming Asterisk
> > Advance co
On Wednesday 17 September 2008 12:02:44 Zeeshan Zakaria wrote:
> I had it setup on rfc2833. Now I've set it up to auto. Will see how it will
> work. But I was thinking is it possible that DTMF tones get distorted on
> their way from my server to the provider's server, which cause this
> problem?
A
On Wednesday 17 September 2008 11:25:36 Zeeshan Zakaria wrote:
> I am having DTMF detection problem on DISA with my callback system. For
> many users, it keeps playing the dialtone even after they have input their
> number. I have trunk setup to both g729 and ulaw. What could be the reason
> for th
On Saturday 13 September 2008 21:14:24 Alex Balashov wrote:
> On Sat, September 13, 2008 10:08 pm, Tilghman Lesher wrote:
> > Basically, it's a way to retrieve multiple values from a database without
> > using ARRAY() to enumerate each field, but also without polluting the
&g
On Friday 12 September 2008 18:31:23 Alex Balashov wrote:
> The short answer is SIP.
>
> Stefan Gofferje wrote:
> > http://www.voip-info.org/wiki-IAX
> > http://www.voip-info.org/wiki-IAX+versus+SIP
> > http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
The longer and more accurate answer is
On Thursday 11 September 2008 15:47:19 Eric Chamberlain wrote:
> On Sep 10, 2008, at 7:11 PM, Tilghman Lesher wrote:
> > On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote:
> >> On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote:
> >>> On Wednesday 10 S
On Thursday 11 September 2008 03:09:19 Julien Claassen wrote:
> Greetings!
>I have got a "systematic" problem with the SHELL function. Consider this
> dialplan snippet:
> *** CUT ***
> exten => NUM,1,Set(myreturn=${SHELL(ast_picker sound_file)})
> exten => NUM,2,Answer()
> exten => NUM,3,GotoIf
On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote:
> On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote:
> > On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote:
> >> Hi to all, I actually have an asterisk server configured to write
> >> CDR mys
On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote:
> Hi to all, I actually have an asterisk server configured to write CDR mysql
> records in the same machine (localhost), but I want to write this records
> to another machine also in mysql at the same time, It is possible? It
> means
On Tuesday 09 September 2008 16:36:37 Dean Collins wrote:
> There have been at least 4 announcements with dates etc, this is really
> just the "last chance reminder email".
It's the first I've seen of it. In any case, if this was the last in a series
of reminders, I'm puzzled why the email starte
On Tuesday 09 September 2008 14:00:00 Ming Yong wrote:
> Voiceroute is proud to announce DruidCON to be held from 1-2 Oct 2008 in
> Atlanta GA so that Asterisk users can join us after Astricon :) DruidCON is
> the premier meeting for Druid community users and developers with interest
> in Unified C
On Monday 08 September 2008 23:00:50 Steve Edwards wrote:
> On Mon, 8 Sep 2008, Tilghman Lesher wrote:
> > On Monday 08 September 2008 12:43:53 Nguyen wrote:
> >> Hi listers,
> >>
> >> We want to implement one call center with asterisk. The idea is it
> &g
On Monday 08 September 2008 12:43:53 Nguyen wrote:
> Hi listers,
>
> We want to implement one call center with asterisk. The idea is it should
> be scalable, with openser as an dispatcher and bunch of asterisk servers to
> do ACD, Queues, Agents things... Easy to say :(
>
> Look closely to the curr
On Monday 08 September 2008 13:11:38 Joseph L. Casale wrote:
> I have an Asterisk server running iptables with a public interface
> and an internal interface. I had to change the subnet of the internal
> interface and now I see messages scrolling "destroying.. 192.168.100.1"
> which is the old of t
On Thursday 04 September 2008 12:59:33 MFH wrote:
> As a result of:
> http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html
>
> I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there
> something I should be doing to address this warning?
>
> [Sep 4 13:31:34] WARNING
On Thursday 04 September 2008 05:55:48 Nhadie wrote:
> Hi,
>
> I am trying to test the ilbc codec on asterisk.
>
> allow=ilbc
> disallow=all
>
> using zoiper on two extensions, i set codec on zoiper to ilbc and
> disabled other codecs
>
> tested a call, looked at the channel:
>
>NativeFormats:
On Thursday 04 September 2008 04:25:27 Chris Bagnall wrote:
> I finally got round to upgrading a few of our 1.2 servers to 1.4 over the
> last few days. Most of the changes in config files went without a hitch,
> but this one bothers me:
>
> ERROR[15836]: config.c:750 process_text_line: Future vers
On Thursday 04 September 2008 03:15:51 selmak se wrote:
> AGI Rx << EXEC Dial SIP/1001,,L(32000:2:1)
>
> [Sep 4 11:04:20] WARNING[18100]: chan_sip.c:2907 create_addr: No such
> host: 1001,,L(32000
The issue is that internally, the application argument delimiter in 1.4 is
actually the pipe
On Saturday 30 August 2008 19:15:36 David Burgess wrote:
> Now we've discovered a new problem: Asterisk lets these non-existent
> make calls even though they are not listed as users in sip.conf. We
> suspect that is happening because they are all localhost connections,
> and therefore bypassing so
On Friday 29 August 2008 09:58:56 Karl Fife wrote:
> Anybody care to muse on Wi-SIP vs. SIP-DECT?
>
> My limited research indicates that none of the WiSip phones will ever be
> able to match the performance of DECT phones. Maybe I'm wrong but a
> Wi-SIP phone seems like a DIESEL sports car. There
On Thursday 28 August 2008 19:59:55 Lee, John (Sydney) wrote:
> > > Better still - is it possible to SSH (or some sort of connection
>
> method)
>
> > > from a remote PC to the Asterisk server and make a call using CLI?
> >
> > Sure, you can use the CLI 'console dial' command.
>
> Do you mean that
On Thursday 28 August 2008 19:05:23 Lee, John (Sydney) wrote:
> >> Hello all!
> >> Is there a way to (mis)use asterisk itself as a softphone? Can
> >> I make a call
> >> from within the CLI? Can asterisk from itself produce a ringtone? I
> >> Or can bind a system-command to incoming calls?
> >> Any
On Thursday 28 August 2008 08:59:29 Geraint Lee wrote:
> On 2008/8/28, Cory Andrews wrote:
>
> > Just a heads up, Hitachi is effectively ceasing production of their
> > IP5000 and IP3000 WiFi SIP Phonesproduct availability is next to nil
> > on these. They also have no plans apparently to conti
On Thursday 28 August 2008 08:06:37 Jaap Winius wrote:
> Are there any reliable wireless SIP phones available on the market yet?
I've gotten a Hitachi WIP3000, which works great. Supports b & g, all the
wireless encryption standards, scans networks, everything a laptop softphone
would do, but in
On Wednesday 27 August 2008 09:29:55 Tilghman Lesher wrote:
> On Wednesday 27 August 2008 08:56:02 J.M. wrote:
> > On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher <
> >
> > [EMAIL PROTECTED]> wrote:
> > > Given that this is the case, we may want to do one of t
On Wednesday 27 August 2008 08:56:02 J.M. wrote:
> On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher <
>
> [EMAIL PROTECTED]> wrote:
> > Given that this is the case, we may want to do one of the following:
> > a) document that qualify=yes is incompatible with realtime, u
On Tuesday 26 August 2008 19:28:17 JR Richardson wrote:
> I received a request for a special application and need some guidance.
> Cust has there own Asterisk PBX with SIP phones, pretty standard
> setup.
>
> They want an after hours application that checks inbound caller ID
> numbers and matches
On Tuesday 26 August 2008 09:08:58 Gordon Henderson wrote:
> On Tue, 26 Aug 2008, Tilghman Lesher wrote:
> > On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
> >> A-Ha... That string is 256 characters long... Now there's a fishy number
> >> if ever there
On Tuesday 26 August 2008 08:06:10 Steve Totaro wrote:
> On Tue, Aug 26, 2008 at 9:01 AM, Tilghman Lesher
>
> <[EMAIL PROTECTED]> wrote:
> > On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
> >> A-Ha... That string is 256 characters long... Now there'
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
> A-Ha... That string is 256 characters long... Now there's a fishy number
> if ever there was one.
>
> So, if this a real limitation? This is 1.2.30 if that makes a
> difference...
>
> Did this limit go away in 1.4 ?
Yes, it did.
--
Til
On Monday 25 August 2008 17:07:18 J.M. wrote:
> > I have also reproduced the problem, albeit on the same machine, by
> > setting the "qualify" field to "yes".
>
> A more complete explanation of how I am able to reproduce the issue is:
>
> 1) Disconnect the client.
> 2) Change the client's record's
On Monday 25 August 2008 07:08:30 Szasz Szabolcs wrote:
> Hi!
> I am running CentOS 5 with Asterisk 1.4.21.2 I am trying to setup storage
> of voicemail messages into MySQL. I installed unixODBC unixODBC-devel
> libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured
> and built A
On Sunday 24 August 2008 14:17:47 Karl Fife wrote:
> For crude IPS/IDS is there an Asterisk method to blacklist registrations
> from a specific IP address after a certain number of failed registration
> attempts, or would I need an SBC or IDS/IPS for that?
There is no solution in Asterisk currentl
On Thursday 21 August 2008 10:08:53 J.M. wrote:
> I am running Asterisk 1.4.21.2 with Realtime. I have a phone setup in the
> database and when I connect that phone to Asterisk there are suddenly an
> endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> 'dynamic'" queries being ru
On Monday 18 August 2008 12:02:15 Vadim Lebedev wrote:
> > > Value-01: >888,1,Noop(888)
> >
> > It may or may not be related, but those greater-than signs (>) in there
> > look wrong, and may be causing some problems.
>
> Without them you can't get "exten=>888,1,Dial(888)"
> if your remove '>'
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