,
not on the server. E.g. the IP in ''[EMAIL PROTECTED] is the IP
of the Polycom phone, not the * box. Is there any way to fix this? Rewrite
sip headers? Any ideas?
-josiah
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.
If anybody wants more info, feel free to ask. I also have the dialtone.gsm
file available - or its easy enough to record your own.
Cheers!
-josiah
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in the initial
distribution.
Then just dial 8500 from your phone, enter the extension as defined in
voicemail.conf and password for that extension (extension is what you put in
when Allison asks for 'mailbox...').
That help at all?
-josiah
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, that sounds easiest to me
-josiah
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of all the
sound files and their transcripts...i just spent about 20 minutes looking for
it in the /usr/src/asterisk CVS tree that I checked out - cant seem to find
it off hand. Any body have any idea what that file is?
-josiah
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On Tuesday 05 April 2005 2:40 pm, MF Hulber wrote:
asterisk/sounds.txt
Josiah Bryan wrote:
On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote:
Hello all,
I am looking for a list of all available sound files for asterisk and a
transcription of their content, so that I can have someone
clean up the
opdial.pl enough for general consumption.
-josiah
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script from
the command line.
-josiah
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at /usr/src/asterisk/sounds.txt, where /usr/src is the location of your
asterisk CVS tree. sounds.txt has both the file name and the transcript of
the audio.
-josiah
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,AGI(pickup.pl)
If anyone is interested in pickup.pl, let me know and I'll see what I can do
to make it available.
Cheers!
-josiah
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On Thursday 07 April 2005 11:20 am, Rich Adamson wrote:
What you are asking for (in US terms) is directed call pickup.
Asterisk does not have a directed call pickup implemented
within it. Not sure how one would try to implement that, but
a guess would be that it would require an
. Write a custom AGI program (any language) to do the MSSQL
query. AGI can control playback of audio, etc. See
http://www.voip-info.org/wiki-Asterisk+AGI.
Cheers HTH -
-josiah
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this for transfering calls for my
receptionist.
So, to answer your question, just parse the output of 'show channel
SIP/401-' and grab the 'Direct Bridge' line. Thats about the easiest that
I know of..
HTH -
-josiah
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may say different, but are they really logical to the common man?
Or even to technical users who dont care about the RFCs and just want to do
their work?
-josiah
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if that
suites your needs.
Or:
Action: Command
Command: database put path/to key value
-josiah
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register sepratly? E.g. is there a seperate entry in
sip.conf for each line or do they both register as the same sip device?
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the call to the next available line on the
IP500 using AGI 'EXEC' to run the 'Dial' app.
If anybody is interested in the script, ill try to clean it up enough to post.
-josiah
- Original Message -
From: Josiah Bryan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
directory from a
central server? That way, all servers share the same spool and the MWI will
get reflected on all servers.
-josiah
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+extension
http://www.voip-info.org/wiki-Asterisk+standard+extensions
http://www.google.com/search?hl=enlr=c2coff=1q=site%3Avoip-info.org+Asterisk+s+extensionbtnG=Search
Google is our friend...
:-)
Thanks
Your welcome.
Cheers!
-josiah
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!
-josiah
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but no musiconhold loaded.
-- AGI Script Executing Application: (Dial) Options: (SIP/201|30)
-- SIP/213-090126f8 is ringing
asterisk*CLI
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).
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anything you need. Have a look at
the dmesg output.
On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan
[EMAIL PROTECTED] wrote:
Hello, folks -
Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the
'asterisk' process. I thought it was due to mpg123 and music on hold -
so I
if the
commands are depreciated and you also understand that a lot has change
on zaptel which is now DAHDI
On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan
[EMAIL PROTECTED] wrote:
Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3
(kernel 2.6.9-1.667). (System
why Dial seems to fall thru after the operator xfers the call
or if I can even do anything about that?
Thanks for your help and your time with all of this.
Cheers!
-josiah
Josiah Bryan wrote:
From was a CVS-HEAD version from way back pre 1.2 days, sometime in
the 1.0 days (I think.)
I've
) exited non-zero on
'SIP/josiah2-09f0ea20'
Really destroying SIP dialog '[EMAIL PROTECTED]'
Method: BYE
asterisk*CLI hangup
No call to hangup up
I'm open to any and all suggestions.
Thanks for your time and patience!
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these questions.
Regards,
-josiah
Andrew Latham wrote:
I know many are thinking this but why don't you use a queue with
fewestcalls for the strategy?
On Wed, Sep 3, 2008 at 4:04 PM, Josiah Bryan
[EMAIL PROTECTED] wrote:
Alright, praise diety, I think I've got an idea on *what* its crashing
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Josiah Bryan wrote:
[paging]
exten = 249,1,Goto(paging,s,1)
exten = s,1,Playback(beep)
exten = s,n,Dial(Console/dsp)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
If the caller has hung up, to whom are you
A simple AGI script would be able to handle that easily, I would think.
Or am I missing something in the details?
-josiah
Sriram wrote:
Hi All
I am a premium IVR content service provider thats runs on premium rate
lines, my setup (currently on PRIs) is like customer dials the short
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PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan
Sent: September 29, 2008 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Knowing incoming call technology and channel
[SOLVED]
So, should we (I can do it, if desired) write
~' block (essentially, a pre block) - formatting can
be applied later if requested - and if presented with a reliable
formatting algorithm.
Let me know what you all think. Cheers!
-josiah
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Tzafrir Cohen wrote:
On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
Hey All -
Link to the index page:
http://www.voip-info.org/wiki/view/Asterisk+Documentation
Why not link to the SVN instead?
I considered that as well. My thoughts:
1) Ungoogleabelness (if thats
Tzafrir Cohen wrote:
On Tue, Sep 30, 2008 at 01:35:00PM -0400, Josiah Bryan wrote:
Tzafrir Cohen wrote:
On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote:
Hey All -
Link to the index page:
http://www.voip-info.org/wiki/view/Asterisk+Documentation
Why not link to the SVN
Rob Hillis wrote:
Josiah Bryan wrote:
Any formatting can be added as desired - this was just a quick way to
get the content online.
Might I suggest including...
print -=NOTE: These pages are automatically updated once per
day/week/month/year/decade from the Asterisk subversion
the SIPAddHeader command given above - doesn't work with
the Polycom Soundpoint IP 500 that I tested with. (Even with the missing
'}' at the end that I fixed - still doesn't work.)
Good idea thought - anybody have any magic that might make that work?
-josiah
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... More ...
True Call Queueing
Depends on: res_monitor(M)
Any ideas on how to figure this out?
Many thanks,
-josiah
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Gotta love flukes - after stopping asterisk and restarting so I could
see the startup text, core show application Queue just worked . ???
Oh well. Thanks!
-josiah
Andres wrote:
Josiah Bryan wrote:
Hey All -
Slight problem here - my install of 1.4.21.2 seems to be missing the
Queue
right now.
Cheers!
-josiah
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sprinkling of AGI...
-josiah
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Is this right after bringing online the alias IP?
If so, you might try using arp-sk to broadcast an ARP packet to
kick-start the IP lookup...
http://sid.rstack.org/arp-sk/
-josiah
Vieri wrote:
I'm trying to figure out how to reload iax2 (without breaking existing calls)
so it can listen on a
Vieri wrote:
--- On Fri, 1/9/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
--- On Fri, 1/9/09, Josiah Bryan
jbr...@productiveconcepts.com wrote:
Is this right after bringing online the alias IP?
If so, you might try using arp-sk to broadcast an
ARP
packet to kick-start the IP
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Tim Panton wrote:
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java
sse sse2 ss ht tm
bogomips: 2924.54
==
Thanks for any help or advice anyone may have. Cheers!
-josiah
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Roderick A. Anderson wrote:
Doug Lytle wrote:
Josiah Bryan wrote:
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
The first place I usually start is with memtest86
Here, here!
Every time I have had problems with a system (not just Asterisk
anyone may have. Cheers!
-josiah
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for another any time now...
-josiah
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Josiah Bryan wrote:
David Gibbons wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell
Doug Lytle wrote:
Josiah Bryan wrote:
Roderick A. Anderson wrote:
How would I go about pinpointing / diagnosing the hardware fault? Not
sure exactly what to do with memtest86 - any pointers?
A lot of distros have memtest86 as a boot option on the CD/DVD. You
select it and let it run
the replacement in everything was still fine.
Wilton
Just the application crashes.
I'll try changing RAM simms to see if that helps.
Thanks!
-josiah
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Paul Chambers wrote:
Josiah Bryan wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell
On Monday 18 April 2005 5:56 pm, Dan Levine wrote:
I forgot the command to have asterisk dial and hangup from the console.
dial
hangup
(try 'help' from the CLI)
-josiah
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' at.
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On Monday 25 April 2005 3:05 pm, Daniel Salama wrote:
I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed with
some additional digit(s). How can this be specified on extensions.conf?
exten =
On Tuesday 26 April 2005 9:58 am, Anton Krall wrote:
Guys.
Im using YAC to send callerid info to PCs and I was wondering if there is a
way to get the IP of a certain SIP or IAX client/technology when a dial
command is issued.
For example, if the dialplan has a dial sip/client or
off list.
-josiah
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Josiah Bryan
|Sent: Martes, 26 de Abril de 2005 09:59 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] YAC and IPs
|
|On Tuesday
On Tuesday 26 April 2005 8:01 pm, William Suffill wrote:
Why not DBGet from the SIP.Registry in ASTDB? Wouldn't that be cleaner
since it would only return the 1 you want instead of parsing what
could be a load of sip peers?
Hmm, great idea! (This is one of those DUH! moments for me) - I never
, it will work with the latest cvs stable
version.
**Please CC me directly on a reply to this, since I sometimes miss replies on
the list**
-josiah
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On Wednesday 27 April 2005 10:40 am, Tony Mountifield wrote:
I've been scratching my head trying to think of a way to do this, but
without success yet.
I'm using the Manager API. If I have two channels linked to each other
(i.e. direct connection), or even if they are independent channels,
I
-1f3c
ChannelB: Zap/6-1
It sounds like the intrinsic functionality for 'bridging' is already there in
Asterisk (duh!), it just needs to be encapsulated in a manager action.
Any takers? Maybe a bounty is needed...?
-josiah
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On Wednesday 27 April 2005 2:02 pm, Paul Shiflet wrote:
I just received my TDM400 card from digium with 2 fxo and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
phones. How do i interface my POTS phones with this; can i just crimp an
RJ45 connection on the end of the
mean, is it possible that its as
easy as: 'void ast_bridge_chan( ast_chan * a, ast_chan *b )' and we just need
to package it as a manager action? Anybody have any pointers on how to
proceed?
-josiah
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On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to my
telco provider with source phone number 55)
Is
.
Anyway, all that just to say be nice and dont melt my serves. :-)
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http
a channel not
initalized, you wont hear anything until the channel is initalized, even if
the audio has already started.
At least, thats my non-developer-ish understanding of the sequence of events
after having the same problem myself...
HTH,
-josiah
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pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 2924.54
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not reproducable manually...
Anybody have any ideas?
-josiah
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, and secret
correctly, when you click on a contact's number, your phone will ring.
Any questions, let me know!
Cheers!
-josiah
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asteriskdial
Description: Perl program
- the linux servers with the modems just do standard routing...
At least, i think thats what your asking..
-josiah
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Asterisk
about france. The 2.5mm headset works fine on the 10+
SPA-841's that I've used it on (several managers use headsets in my office
with the SPA-841.)
-josiah
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)
exten = *,1,AGI(checkvm.pl|${ARG1})
exten = *,n,Goto(mainmenu-restart,s,1)
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the top-of-the-head thoughts.
-josiah
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, -2x, -3x stuff? Dunno bout those...since the SPA-841 hotline
trick is just a special dialplan command, i think it might work on the
SPA-[123]x models, but i havn't tried it yet.
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I'd vote for DNS problem myself. Do you have a local dns sevre that forwards
unknow requests?
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-Original Message-
From: Danny Dias ing.diasda...@gmail.com
Date: Sun, 7 Feb
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