Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On Thu, Mar 16, 2006 at 04:27:10PM -0700, Douglas Garstang wrote:
Thanks for the reply, but still no luck.
sip.conf:
[2944093]
...
regcontext=sip_autoreg
David,
How's DUNDi make this redundant? The way I understand it, a phone is only ever
registered to a single Asterisk box at a time. If that Asterisk box where to
fail, callers lose the ability to contact users that where registered on that
box.
-Original Message-
Grrr. I'm using outlook web access and there's no way to do inline replies.
Anyway...
Gabriel.
Using SER does not create a single point of failure. You install three SER
boxes. Single point of failure gone.
It does not take several seconds.
If your phones are configured for SRV, and 2/3 of
Gabe,
I'm the guy with the three SER boxes. All three SER boxes are active. Polycom
phones point to a DNS domain with SRV lookups for all outbound requests. The
SRV queries return the three SER boxes. The three boxes authenticate a user and
then store the user in it's 'location' database
: [Asterisk-Users] Feedback from VON expo! Info on * HA
andPolycomphone!!
On 17/03/06, Douglas Garstang [EMAIL PROTECTED] wrote:
David,
How's DUNDi make this redundant? The way I understand it, a phone is
only ever registered to a single Asterisk
Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On Thu, Mar 16, 2006 at 04:41:42PM -0700, Douglas Garstang wrote:
-Original Message-
From: Luigi Rizzo [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 16
.
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Thu 3/16/2006 11:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On 17/03/06, Douglas
The phone must have transported you to Australia... :)
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 10:05 AM
To: asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Double-ring tone
I upgraded my
You could use Asterisk to flip the caller id and dnis variables, before you
Dial the phone. :)
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Why? If you flip the callerid and dnis variables, it should work with any phone.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] how to
Aren't you bothered by the fact that the sound file quality goes up and down as
different sound files are played? It's quite obvious to hear the difference
between a ulaw file and a gsm file.
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15,
Is anyone using realtime sip for friends (ie phones) with multiple Asterisk
boxes all pointing to the one central MySQL database? Does it work? Are phones
that are registered to the database from Asterisk box able to reach phones
registered to the database from another Asterisk box?
Doug.
Is it
just me or is the voip-info web site down right now? Jeez that web site is
flaky.
___
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To UNSUBSCRIBE or update options visit:
: [Asterisk-Users] Voip-Info
On Tue, 2006-03-14 at 10:21 -0700, Douglas Garstang wrote:
Is it just me or is the voip-info web site down right now? Jeez that
web site is flaky.
Is it just me or was this message in HTML? Jeez some people never learn.
Dave Cotton [EMAIL PROTECTED
Does
anyone know if realtime extensions allows extensions in the format
callerid/extension yet? ie the extensions.conf file allows you to
do:
5551212/1000 = exten ...
and it
matches against extension 1000 when the caller id is 5551212. Last time I
checked, realtime didn't support this
for this feature?,
perhaps something similar can be achieved in a different manner.
On 3/14/06, Douglas
Garstang
[EMAIL PROTECTED] wrote:
Does anyone know if
realtime extensions allows extensions in the format callerid/extension yet?
ie the extensions.conf file allows you to do
program it into the
realtime extensions... maybe using gotoif's?
Aaron
Douglas Garstang wrote:
Uhm caller id based routing?
-Original Message-
*From:* Sig Lange [mailto:[EMAIL PROTECTED]
*Sent:* Tuesday, March 14, 2006 4:33 PM
*To:* Asterisk Users Mailing List - Non
Boy, am I stuck...
I'm officially ready to toss Asterisk out the window. I have to admit it isn't
necessarily all the fault of Asterisk either. It just seems that every option I
turn to suddenly ends in failure. I don't know if it's me that's bitten of more
than I can chew with this project,
I discussed the native sounds with my boss the other day. We decided not to use
them because there's only 197 sound files out of a total of 1200 installed on
the system from asterisk and the asterisk-sounds packages. We wanted to have a
consistent playback quality, rather than have it going up
Discussion
Subject: Re: [Asterisk-Users] Clustering
On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote:
Kevin,
From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext:
If regcontext is specified, Asterisk will dynamically create and destroy a
NoOp priority 1
the registration)
process the call?
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Monday, March 13, 2006 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clustering
On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Thanks
: Re: [Asterisk-Users] Clustering
On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote:
Thanks Kristian. It isn't clear how this means a registration on one Asterisk
system magically appear on the other though...
I'm not quite certain as I build my call routing
on scripts instead
I'm not the person who your replying to, but I'll jump in anyway. We're using
MySQL in conjunction with AGI. This gives us the ultimate flexibility (and me
the most freekin work).
Databases are configured in a HA manner. Not sure exactly what form that will
take yet, but will be either
Holy crap. You got SIP realtime working? I've tried it twice before and it
failed the same way twice. Do you have multiple Asterisk boxes accessing the
same sip info (ie phones) in the same table on the same database? Digium has
said numerous times this known not to work, although I cant' work
he does not know your setup) start
sending RTP packets to the other interface?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Douglas
Garstang
Sent: Sunday, March 12, 2006 1:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users
then using regexten on all servers so
when a * tries to make a call it can find where to go, or are you using
something else?Thanks!Ron
On 3/12/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
It
doesn't. It's transparent to the user agent.-Original
Message-From: Wai Wu
Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering / Dundi
11 mar 2006 kl. 07.54 skrev Douglas Garstang:
Hi JR. I'm dying to know... where'd you find your DUNDi
documentation? Has something new appeared since I looked
-
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; Asterisk Users Mailing
List -Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, March 10, 2006 10:49 PM
That's what we're using. :)
-Original Message-
From: David Coulson [mailto:[EMAIL PROTECTED]
Sent: Sat 3/11/2006 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering
No, only if a network interface in the server fails. We have two network
interfaces per system (actually we have four, but two are on a private network
with a MySQL server). If one of the network interfaces fails, OSPF will switch
the default route over to the other interface pretty quick
If you
implement multiple Asterisk systems, your challenge is going to be in ensuring
that phones registered to one Asterisk know how to reach phones registered to
another Asterisk system. Good luck with that!
Doug.
-Original Message-From: Ron McCarthy
[mailto:[EMAIL
with an internal DNS zone to
lookup the routing information. SIP phone logs into Asterisk 'A' and
a script runs to update the e.164 DNS info pointing the DID to
Asterisk 'A'
-Matt
On Mar 10, 2006, at 2:42 PM, Douglas Garstang wrote:
If you implement multiple Asterisk systems, your challenge is going
, March 10, 2006 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clustering
Douglas Garstang wrote:
I think there's a difference between sharing extension info and sharing
registration info. Just because an extension exists on a given Asterisk box
PROTECTED]
Sent: Fri 3/10/2006 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering
Douglas Garstang wrote:
I'd just die to see an example of that. I've never
We're doing this. Our Polycom phones point to a domain name that support SRV
records which gives us a roughly even distribution of calls. We have OpenSER
systems sitting in front of the phones. Each OpenSER system is configured with
different primary/secondary/tertiary Asterisk boxes. When a
Hi JR. I'm dying to know... where'd you find your DUNDi documentation? Has
something new appeared since I looked at it 2-3 months ago? The O'Reilly book's
DUNDi section was impossible to follow, and the examples in the Asterisk DUNDi
config files are no better. You do a search online and get
Good grief! I posted the message below at 1:17pm... and it appeared on the list
after 8pm.
Nice
-Original Message-
From: Douglas Garstang
Sent: Tue 3/7/2006 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc
This is a SER/Polycom question, but I hoped we may have some SER guru's here...
I have a series of Polycom phones that are tying to register with OpenSER. The
phone sends a REGISTER message, and OpenSER replies with Unauthorised (all
normal). The phone re-sends the REGISTER with the
Docs? Polycom has docs? Where would one find this fabled land of... err I mean
Polycom does stating what ftp servers are supported?
Doug.
-Original Message-
From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 12:12 PM
To: Asterisk Users Mailing List -
Asterisk
what it is today.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Tuesday, March 07, 2006 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Oh this is bad
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic
Douglas Garstang wrote:
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure
seems to be missing a few features.
Um...it's Open Source. Why don't you add
:[EMAIL PROTECTED]
Sent: Wed 3/8/2006 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic
Douglas Garstang wrote:
Asterisk calls
Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] res_mysql.conf DNS SRV lookup
Douglas Garstang wrote:
Good grief! I posted the message below at 1:17pm... and it appeared
on the list after 8pm.
Nice
I have
a configuration where RTP traffic is going out interface pub0, and coming back
into through pub1.
I have
bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows:
udp 0 788
0.0.0.0:5060
0.0.0.0:*
which
means that Asterisk is listening on all addresses (on all
interfaces?).
Anyway,
this is bad bindaddr and rtp traffic
On Tue, 7 Mar 2006 09:12:25 -0700
Douglas Garstang [EMAIL PROTECTED] wrote:
I have a configuration where RTP traffic is going out
interface pub0, and coming back into through pub1.
I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
shows:
udp
Are
you kidding? Asterisk doesn't do SRV. If you read all the VOIP books out there,
SRV lookups are _the_ way to achieve redundancy. Digium hasn't gotten to it I
guess with their 'enterprise class' product though.
-Original Message-From: Ramin Nikaeen
[mailto:[EMAIL
: [Asterisk-Users] res_mysql.conf DNS SRV lookup
7 mar 2006 kl. 18.12 skrev Douglas Garstang:
Are you kidding? Asterisk doesn't do SRV. If you read all the VOIP
books out there, SRV lookups are _the_ way to achieve redundancy.
Digium hasn't gotten to it I guess with their 'enterprise class
Yay!
-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] res_mysql.conf DNS SRV lookup
7 mar 2006 kl. 19.03 skrev Douglas Garstang:
My bad
ethernet
The IP300/301 has the power jack, the IP500/501 the inline cable.
PaulH
On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
Not true. Some do and some don't. Some have a place to plug a separate DC
adapter, and some have the inline power, where the adapter plugs into the
ethernet
I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to see what's happening.
self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd(background,enter-conf-call-number,)
self.agi.execCmd(Read,confNum|||,)
from 216.188.128.11:5060:
REFER sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
CSeq: 2 REFER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL
I have the following in extensions.conf:
exten = 1000,1,Meetme(|dMic|)
According to the 'show application meetme' docs:
'i' - announce user join/leave (new in Asterisk 1.2)
Well, when users join the conference, Asterisk records their name, but does not
broadcast it into the conference. I
Anyone seen this? If not I guess I'll have to post it as a bug.
Extensions.conf has this:
exten = 123,1,Meetme(|dMic|)
I dial 123, and enter my conference number. Asterisk asks me to enter my name.
At this point I hang up. If I type at the Asterisk console 'meetme list 12345'
it shows that I
Why do you need to have to set incominglimit=1 for buddies to work? We've not
had that requirement.
Doug.
-Original Message-
From: rivy strauss [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Buddy watch?
Hi,
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme Participant Announcement
Douglas Garstang wrote:
I have the following in extensions.conf:
exten = 1000,1,Meetme(|dMic|)
According to the 'show application meetme' docs:
'i' - announce user join
Not true. Some do and some don't. Some have a place to plug a separate DC
adapter, and some have the inline power, where the adapter plugs into the
ethernet cable. Not sure which ones are newer, and which are older.
-Original Message-
From: Michael Welter [mailto:[EMAIL
Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk at large
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
The best way to achieve maximum manageability is to design a MySQL database
and develop AGI scripts (in your language of choice) that work to that
design. I've found
from 216.188.128.11:5060:
REFER sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
CSeq: 2 REFER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL
I have the following in extensions.conf:
exten = 1000,1,Meetme(|dMic|)
According to the 'show application meetme' docs:
'i' - announce user join/leave (new in Asterisk 1.2)
Well, when users join the conference, Asterisk records their name, but does not
broadcast it into the conference. I
I have ztdummy installed:
Module Size Used by
ztdummy 3464 0
zaptel218756 1 ztdummy
crc_ccitt 2176 1 zaptel
ohci_hcd 16388 0
floppy 49028 0
pcspkr 2180 0
piix
The file is at /var/lib/asterisk/astdb
-Original Message-
From: Joseph Tanner [mailto:[EMAIL PROTECTED]
Sent: Friday, March 03, 2006 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Does an entry in AstDB stay after reboot?
Yes, the AstDB
I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to see what's happening.
self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd(background,enter-conf-call-number,)
self.agi.execCmd(Read,confNum|||,)
Yikes. Managability! It's a lot easier to manage multiple Asterisk systems
configuration from a single MySQL database then it is to manage .conf files on
several redundant Asterisk boxes. I can't believe you asked that question. I'll
apologise in advance because I must be missing part of this
.
Also, what is the purpose of NOT having *any* configs from
/etc/asterisk/
On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Yikes. Managability! It's a lot easier to manage multiple Asterisk systems
configuration from a single MySQL database then it is to manage .conf files
, Douglas Garstang [EMAIL PROTECTED] wrote:
Yikes. Managability! It's a lot easier to manage multiple Asterisk
systems configuration from a single MySQL database then it is to manage
.conf files on several redundant Asterisk boxes. I can't believe you asked
that question. I'll apologise in advance
Marco,
Which versions of Asterisk will that patch work with?
Douglas.
-Original Message-
From: Marco Maiolini [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 6:36 AM
To: asterisk-users
Subject: Re: [Asterisk-Users] BLF not working after reload
I solved that problem for
I'm trying to find a way in
extensions.conf to match ANYTHING dialled, including characters such as
*.
The following works for
numbers...
exten =
_X.,1,AGI(script)
but doesn't catch when someone dialls *
first. I tried this:
exten =
_.,1,AGI(script)
which catches when someone dials
I'd like to use the AGI command CHANNEL STATUS to check the status of a
channel. However, the dial() command doesn't return -1 until after the call has
hung up. If that's the case, how is channel status supposed to return statuses
like:
status values:
0 Channel is down and available
1
MC,
But the channel status command is documented as an AGI command itself.
If you look at http://www.voip-info.org/wiki-Asterisk+AGI, you'll see the
'channel status' command listed there as an AGI command.
I can't post my dial plan, as I don't really have one. Well, I do, and it looks
MC,
I think I worked out that I need to use ${DIALSTATUS} anyway. Don't really see
what 'channel status' is for...
-Original Message-
From: Douglas Garstang
Sent: Monday, February 27, 2006 1:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk
Ok, here's a weird one.
I have an AGI script where one user calls another. The call is answered.
Everything is peachy. If the call is terminated by the CALLEE hanging up the
call, then Asterisk returns control back to where the Dial() command left off,
and I can check the return code of
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI Scripts Terminate too Soon
Douglas Garstang schrieb:
...
HOWEVER, if the CALLER hangs up the call, it seems
Hi,
did you try the dial command option g?
I did not neither, but when I understand
If that's true, why does dial() return control to the script when the callee
hangs up?
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AGI
If you do a 'reload' in Asterisk, it deletes all the sip subscriptions. Do a
'sip show subscriptions' before and after a reload command. They will
disappear. I've been bitching about this for a while, and asking why
subscriptions can't be stored in astdb like registrations.
If you reboot the
Polycom does support Asterisk, Asterisk Business Edition.
-Original Message-
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What business IP phone to use
You don't need the Polycom ACD support in order to do ACD logins with Polycom
phones. Just dial an extension and call AgentCallBackLogin(). You won't get any
visual confirmation on the phone however of being logged in, but you will be.
If you set the acd-login fields in the phone's xml, the
This
has worked for several months for us. It's /etc/dhcpd.conf
ddns-update-style ad-hoc;
authoritative;option option-66 code 66 = string;
subnet
172.32.16.0 netmask 255.255.255.0
{ #range 192.168.10.101
192.168.10.120; default-lease-time
600; max-lease-time
7200; option
option-66
is correct.
Thanks,
Doug
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Wednesday, February 22, 2006 4:18 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: RE: [Asterisk
Polycom only supports Asterisk Business Edition. Does ABE even support
hints/buddies?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After
several hours jerking around with icecast and muse, I tried to point my
asterisk system directly at two streams I know work.
This is what extensions.conf has:
[default]
mode=quietmp3
extensions.conf:
exten = 1234,1,Answer
exten = 1234,2,MusicOnHold(stream2)
exten = 1234,3,Hangup
On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote:
Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work.
After several hours jerking around with icecast and muse, I tried
You should be able to throw an OpenSER box in between the two Asterisk systems,
and with a bit of configuration get it working. We do something similar, except
the SER box sits between the phones and Asterisk. It passes the
SUBSCRIBE/NOTIFY messages backwards and forwards between the two.
I have the following in extensions.conf
[incoming]
exten = _X.,1,AGI(python2/iptrouter.py|OnNetOutgoing)
exten = _X.,2,Hangup()
Works well. I wanted to also catch numbers prefixed with a star code, say *,
with the lines above. However, when I dial a star code prefixed number,
Asterisk first
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten =
Douglas Garstang wrote:
It was trying to perform looping in the dialplan that made me
seriously look at AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
Discussion
Cc:
Subject: Re: [Asterisk-Users] is there a web interface to this mailing
list?
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
Too funny. But true.
The first time I did
I'm getting an error back from an AGI Dial command. Weird thing is that it's
STILL performing the Dial.
Here's what I am sending (without the paranthesis):
(EXEC DIAL SIP/1|5|tr)
and here's what I am getting (without the paranthesis):
(510 Invalid or unknown command)
Why would I get this
)
-- Called 1
-- SIP/1-486d is ringing
-- Nobody picked up in 5000 ms
-- AGI Script python/iptrouter.py completed, returning 0
== Auto fallthrough, channel 'SIP/3250072-c5d3' status is 'NOANSWER'
-Original Message-
From: Douglas Garstang
Sent: Thursday, February
Well, I'm about ready to throw Asterisk across the room.
Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI,
if the callee hangs up the call, Asterisk sends a return code, but if the
caller hangs up, it does not???
This means if an agi script services a call, and
:
[Asterisk-Users] AGI Flakyness *sigh*
From:
Douglas Garstang [EMAIL PROTECTED]
Date:
Thu, 16 Feb 2006 09:24:26 -0700
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Well, I'm about ready to throw Asterisk across the room.
Can someone tell me WHY
, Douglas Garstang wrote:
As you can quite clearly see, Asterisk sends no return code back to the AGI
script. I really want to understand why this happens. I also don't like the
Are you using AGI() or DeadAGI() ?
until after the call is Hung up. Also, another annoyance. Why can't
asterisk send
What's the best way to increment a numeric variable in the dial plan?
I tried this...
exten = s,1,Set(mainLoop=${MATH(${mainLoop}+1)})
but that converts it to a floating point number (WHY???), so I end up with
1., which later on means I have to perform string manipulation to get rid
of the
Freddi,
I started out this morning try to proof the concept of having Asterisk call an
AGI script, set several variables, and then return control to the dialplan
where it would execute the command. I wanted to set a number of variables in
the AGI for each number to dial. The first variable
Yes, programming the dialplan is akin to programming assembler.
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 15, 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] is there a web interface
I've got several issues with AGI/FastAGI
1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk
block and not return a result until the command is complete? Specifically, the
dial command. If I send a Dial command to Asterisk, I don't get a return result
until AFTER the
Hi Freddi
Thanks for the reply. Neat ideas there, but a couple of issues.
1. Don't want to have to jump around between the FastAGI and the dial plan. Our
plan is to have NO customer data in the dialplan, as all data will be contained
within MySQL. We don't want to have to make _any_ edits to
So, I'm noticing that when Asterisk executes an AGI script, that the AGI script
keeps running until the call is complete.
Is there any way to have the script terminate when the call is answered?
Also noticed that when user makes a call to user B, if user B hangs up the
call, then Asterisk
I've noticed that Asterisk AGI scripts don't terminate when a call is answered.
Does anyone know how to do this? I would think that this would be a very big
problem, if the scripts stayed in memory, doing nothing, until the call
terminates.
Not only do you have to have a process for routing
that, otherwise you
can sit
there the whole time and handle dial yourself.
All except one of our apps now use DIALCMD and has cut our system load
by 75%
--
~Andy Brezinsky
On Monday 13 February 2006 11:26 pm, Douglas Garstang wrote
!
Douglas Garstang wrote:
Thanks for the reply Kristian, but you've completely confused me.
Asterisk-sounds is the default set of sounds on digium's website?
No. The default sounds are in the Asterisk distribution itself. The
asterisk-sounds package is separate, and none of the built
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