one have suggestions for rational sounding dialtone and ring
indication for North America?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
On Sunday 29 January 2012 8:27:30 am Olivier wrote:
> 2012/1/29 Mike Diehl
>
> > Hi all,
> >
> > I'm working with the Digium fax for Asterisk product, which is working
> > pretty
> > reliably for me.
> >
> > However, the sendfax
I run asterisk from inittab. So, I'd have to create a shell script to do the
ulimit, and then start asterisk. Is there any reason NOT to launch a shell
script from inittab?
On Thursday 26 January 2012 3:53:42 pm Chad Wallace wrote:
> On Thu, 26 Jan 2012 10:35:14 -0700
>
> Mik
;m expecting to get at least FaxStatus, SendFAXStatus and FaxDocument events.
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Joi
Hi all,
While trying to track down a T.38 issue, I came across a series of log
entries like this:
[Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr:
Unable to allocate socket: Too many open files
[
ly seating the handset in the cradle.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Tuesday, January 17, 2012 12:24 PM
> To: Asterisk Users Mailing List - Non
ng any out-bound
issues.
I don't see anything in the logs, and the phones remain registered.
They can't reproduce the symptoms on demand, but it seems like it's happening
more frequently, lately.
Any ideas?
--
Take care
Good Luck!!
>
> Thanks!!
>
> Jim.
>
> - Original message -
>
> > Mike Diehl wrote:
> > > Usually, it just works...
> > >
> > > Any ideas?
> >
> > I've seen this before.
> >
> > One of our facilities have 'smart or
re, so I'm at a loss as to how to proceed.
Usually, it just works...
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us fo
On Monday 12 December 2011 6:39:34 am Barry L. Kline wrote:
> On 12/11/2011 10:59 PM, Mike Diehl wrote:
> > Should I go to 1.8.x? Or all the way up to 10.x? This is a
> > production system and I can't afford to be "testing" code.
>
> The 1.8 series is the c
ast update.
I ended up making some of my updates in the hang-up phase via the "h"
extension. See if that will do what you need.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by h
m: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Monday, December 12, 2011 5:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Multiple route failover zaps reg
;t afford to be "testing" code.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
vailable during failover?
I'm considering using the Tinc VPN solution to prevent the IP address from
chaing, but I'm hoping for a more simple solution.
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- B
On Friday 09 December 2011 12:24:02 am Jeremy Kister wrote:
> On 12/9/2011 12:55 AM, Mike Diehl wrote:
> > What am I doing wrong?
>
> perhaps try: http://jeremy.kister.net/code/asterisk/mwi.pl
I got this to work on the same LAN, but it will not traverse a NAT. At least
it wo
sterisk/voicemail/CONTEXT/USER/busy.WAV'));
I assume this has to be done for every CONTEXT and every USER, and those
values need to be substituted into both the fields, and the directory path
value.
===
Finally, is there an AGI comman
What I get as output is:
(type: 3, code: 3): from phone_ip
A sniffer running on my phone's gateway router tells me that it's sending an
ICMP port unreachable message back to the sip server.
What am I doing wrong?
TIA,
--
Take care and have fun,
Mike Dieh
; -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Thursday, November 10, 2011 3:05 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Frequent Asterisk Restarts
>
t:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
bility? BTW, both servers are running
Asterisk 1.6.2.9 w/ realtime sip/voicemail.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a l
e-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Thursday, October 13, 2011 12:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Phones flapping wit
as
anyone fixed it? Any ideas, otherwise?
TIA.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every T
gt;
> > I'm not able to figure out what causes this and usually, I have to
> > prune the client, or reload sip to make it stop.
> >
> > Can someone tell me what causes it and how to fix it?
>
> An Asterisk dynamic string (struct ast_str) needed more roo
===
I'm not able to figure out what causes this and usually, I have to prune the
client, or reload sip to make it stop.
Can someone tell me what causes it and how to fix it?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
ot;features show" ? on CLI ?
>
> On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl wrote:
> > Hi all.
> >
> > I could have sworn this working at one time...
> >
> > But it doesn't look like any of the functions provided by features.so is
> >
looks sane.
What else should I try?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar e
On Tuesday 30 August 2011 7:12:54 pm Michael L. Young wrote:
> - Original Message -
>
> > From: "Mike Diehl"
> > To: asterisk-users@lists.digium.com
> > Sent: Tuesday, August 30, 2011 5:13:22 PM
> > Subject: Re: [asterisk-users] Polycoms rebooti
> resetting the PoE module which causes your phone reboots.
All of the phones are AC powered. Either via an injector or wall outlet; I
don't remember which. Definitely NOT POE.
--
Take care and have fun,
Mike Diehl.
--
__
e culprit in both cases
> was the network connection. Replacing the cat5 cable to the phone or
> changing the attached port fixed it both times.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
On Wednesday 17 August 2011 4:39:14 pm Andrew Latham wrote:
> On Wed, Aug 17, 2011 at 6:35 PM, Mike Diehl wrote:
> > On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote:
> >> Mike Diehl wrote:
> >> > Any other ideas?
> >>
> >> They should
On Wednesday 17 August 2011 4:11:32 pm Andrew Latham wrote:
> On Wed, Aug 17, 2011 at 6:01 PM, Mike Diehl wrote:
> > Hi all,
> >
> > I've got a customer with 10 Polycom 335's and the latest(ish) firmware.
> > For the most part, things are working well.
On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote:
> Mike Diehl wrote:
> > Any other ideas?
>
> They should be writing out logs to your ftp server (If your provisioning
> them that way).
At the moment, my web server isn't capable of receiving the phones POST
reque
I've got a tcpdump running against one of the phones on my server, but so far,
it's not rebooted, so I've got nothing to look at.
Any other ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwi
A9:B6][501][962522A][0101062C] 000B821CA9B6-1 SIP registration
failed.
Retrying in 20 seconds. Server: 209.250.31.96
===
On Friday 22 July 2011 2:38:15 am Mike Diehl wrote:
> On Friday 22 July 2011 1:42:33 a
On Friday 22 July 2011 1:42:33 am Ishfaq Malik wrote:
> On Thu, 2011-07-21 at 17:13 -0600, Mike Diehl wrote:
> > Hi all,
> >
> > I've got a strange problem with a customer's phones.
> >
> > They've got a bunch of Grandstreams that seem to be rock
gt;
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > Mike Diehl
> > Sent: Friday, 22 July 2011 10:50 a.m.
> > To: asterisk-users@lists.digium.com
>
s go bad. Only
some of the phone lines go down and they stay down until the phone is
rebooted.
I'm not even sure what to look for when I go to the site. Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Band
Hi all,
I'm trying to figure out how it is that a couple lines on a given phone, with 3
lines, can qualify as unavailable while the remaining lines can be available.
I've got qualify=1000 in my sip.cfg.
Shouldn't this be an all-or-nothing proposition?
--
Take care and have
Hi all,
I've got a number of Grandstream phones and I'd like to be able to reboot them
remotely, as I do my Polycoms...
I've got this in my sip_notify.cfg:
[grandstream-check-cfg]
Event=>sys-control
Doesn't seem to work. Any ideas?
--
Take care a
ou able to do that?
I'm scheduling a SIP reload tonight. Might as well do an Asterisk restart
instead.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Ne
, and the
database indicate that lines 2 and 3 are registered.
0004F2127F60-1 is the registration name for line 1 and 0004F2127F60-3 is for
line 3.
Any ideas on how to fix this? Would doing a factory reset and
reprovisioning on the phone help? Or would that be just wheelspin?
TIA
On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote:
> On 06/16/2011 07:58 AM, Mike Diehl wrote:
> > Well, I ran a simple test by trying to configure the second port to use
> > the DNS SRV record, as described below.
> >
> > Here
for production? Should I
move to 1.8, yet?
So far, the only work-around I've come up with is a separate context for EACH
sip account, with the variables hard-wired FUGLY!
--
Take care and have fun,
Mike Diehl.
--
__
diehlnet.com,
which does exist. It should be looking for the SRV record.
What am I missing?
Mike.
On Tuesday 14 June 2011 3:08:33 am Paul Hayes wrote:
> On 13/06/11 19:44, Mike Diehl wrote:
> > Hi all,
> >
> > I'm trying to provision my PAP2T's to use a SVR lookup
records are in place because my Polycom and Grandstream servers work
just fine.
What else do I need to do to get the PAP to work this way?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provid
n
the Polycoms. The sip database records are virtually identical, btw.
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a li
Hi all,
Is it possible to store voicemail in a Mysql database without using ODBC?
I've got RTA sip and voicemail working; I just want to store the messages in
the db now. Configuring ODBC seems like a lot of work if I don't have to.
TIA.
--
Take care and have fun,
a door box (E20 iirc) that is powered by an
> analog line and can do auto answer when it gets the first ring.
I'll look into it. Thank you.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Prov
Hi all,
Is it possible to send a SIP header to a PAP2T or SPA and cause the device
to automatically answer? I can do this with my Polycom phones and would like
to do it with my ATA's.
Any ideas?
--
Take care and have fun,
Mike
fer is complete
xferfailsound = beeperr; to indicate a failed transfer
However, it seems that transfer is a function of the phone, not Asterisk. Is
there any way I can configure the Polycom phones to either use the Asterisk
function, or to make a "beep" when a transfer completes?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
t;FAX COMPLETE",1);
I never see the "FAX COMPLETE" message on the console, I've set verbose to
25. Any ideas? I'd like to take the next few instructions to log
success/failure.
--
Take care and have fun,
Mike Diehl.
--
erisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and C
The only
configured codec is u711.
When the user tries to send a fax, it gets to the point where it issues a
reInvite to start the T.38, then the called side receives a SIP 488 (Not
Acceptable Here)
Where should I start? Any pointers would be most welcome.
--
Take care and have fun,
"Steve Edwards" wrote:
> On Mon, 31 Jan 2011, Mike Diehl wrote:
>
>> I've got an agi script that calls the directory function, which seems to
>> work to a point. However, once the caller has selected an entry, I need
>> my agi script to find out which e
s a variable set, as far as I can
see.
Is there a way to get this information from the directory application?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
N
PSTN connection and replace all of the
office phones.
With these short distances, will I need to worry about echo? Do these
devices have echo cancellation?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colo
or over, say, 5
minutes. We'd like to be able to start our message as soon as the greeting
is done.
Any suggestions?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-d
Hi all,
I've got two questions about CDR's.
1. I'd like to start logging the IP address that a call orginates from.
I'm sure I can get this into the userfield of my CDR table, but what
variable should I use to get this value? I looked at the variables page at
voip-info and didn't find anything
d possible SIP ALGs that are between the devices.
> Check for UDP port forwarding settings, and check that the RTP ports that
> have been negotiated for the call are not conflicting with those of other
> devices/calls/port forwardi
1:47:12 pm Olivier wrote:
> 2010/10/26 Mike Diehl
>
> > Hi all,
> >
> > I seem to be having a strange problem with a sip trunk.
> >
> > On a fairly frequent basis, I'll make a call, ore receive a call, and
> > there will be NO sound. The strange part
he packets
simply aren't being sent.
It only seems to happen on a particular trunk. The same phone calling on a
different trunk works just fine.
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colo
propriate.
Thank you. I should be able to get it working from here.
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
le to get a value for ${id}.
Is this a known limitation, or am I doing something wrong? If this won't
work, is there a work-around?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by
Never mind...
I mistakenly interpreted codec_a_mu.so as some sort of universal translator
between ulaw, alaw, and slin. When I loaded the "rest" of the modules, it
worked like a champ.
Mike.
On Sunday 10 October 2010 6:57:30 pm Mike Diehl wrote:
> I'm doing some final
On Monday 11 October 2010 4:34:46 am Stefan Tichy wrote:
> On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote:
> > Asterisk replied:
> >
> > Peer test not found.
> >
> > So it looks like I'm missing something pretty basic.
>
> I would sugges
I'm assuming use=0 because the server is idle.
I've got allow = all in my sip.conf file.
Anyway, does anyone have an idea on how to resolve this?
--
Take care and have fun,
Mike Diehl.
--
_
accountname is the name of the
> extensions you have to configure.
>
> BR
>
> - Andrea
In the database, I changed the accountname to "test" and tried:
sip show peer test load
Asterisk replied:
Peer test not found.
So it looks like I'm missing something prett
peer | password |
+++-+--+--+
1 row in set (0.00 sec)
However, when I try to get my Ekiga client to register, Asterisk displays
"No matching peer found" on the console.
What should I do next to try to diagnose/fix this?
--
Take car
hat am I doing wrong?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
htt
On Monday 29 March 2010 10:15:50 am jon pounder wrote:
> Mike Diehl wrote:
> > Hi all,
> >
> > I've cross-posted this to the -users and -biz groups. Hope that's OK.
> >
> > I have a customer who REALLY needs to be able to send/receive faxes
> >
one let me know?
Otherwise, is there a product/service they can buy that will allow them to fax
to/from their computers?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.
option boot-server "ftp://polycom:pas...@10.0.1.1";;
option tftp-server-name "ftp://polycom:pas...@10.0.1.1";;
option time-offset -25200;
host 0004f2278ff8 {
hardware ethernet 00:04:F2:27:8F:F8;
}
host 0004f22afafd{
quot;ftp://polycom:pas...@10.0.1.1";;
To be honest, I'm not sure which one is needed, but together, they get the job
done.
Let me know if you need more help.
I owe you one, btw, because I read your blog on getting these beasts
provi
David Backeberg wrote:
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the secur
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
extension that answers the call and runs the musiconhold command with
the appropriate class name.
All I get on the phone is silence. The console tells me that moh
started and immediately stopped, but it complains tha
seen this happen once, but I've been unable to reproduce it reliably.
Any ideas?
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or updat
pe="1"
reg.2.lineKeys="1"
reg.2.callsPerLineKey="3"
I've uploaded this config to the phone, but the symptoms persist. Any clues?
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and
e symptoms before? Any clues as to how to fix it?
TIA,
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update opt
.1.1";;
But what is the equivelent configuration for dnsmasq?
If I can't get this working, I'll have to resort to hard-coding the
information into each of 12 phones.... Yuck!
--
Take care and have fun,
Mike Diehl.
___
-- Bandwidth and
from a
regular call. Can this be done?
--
Take care and have fun,
Mike Diehl.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/m
wn and santized version of the config files available
at:
http://www.diehlnet.com/diehlnet.txt
http://www.diehlnet.com/Polycom-0004f211d1d0.txt
I've changed the extensions on the website from .cfg to .txt so that it will
open better for y
On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
> Mike Diehl wrote:
> > On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
> >> Mike Diehl wrote:
> >>> On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
> >>&g
7;t upgrade very often. When I can/do, I need to
get as much bang for my buck as I can.
--
Take care and have fun,
Mike Diehl.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE o
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
> Mike Diehl wrote:
> > On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
> >> Mike Diehl wrote:
> >>> On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
> >>>> Hi
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
> Mike Diehl wrote:
> > The phone is a Polycom 501; it's been discontinued. I am working on a
> > testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
> > hesitant to upgrade a sys
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
> Mike Diehl wrote:
> > On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
> >> Hi Mike -
> >>
> >>> I've got a Polycom 501 that's been working with Asterisk for some time.
>
is the case, but I've attached a copy of my phone.cfg file,
if you want to have a look. The passwords have been sanitized
ere to start to fix it?
TIA,
--
Take care and have fun,
Mike Diehl.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/li
ycontext to the name of the context that you want to
> use.
>
> That may work for you.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Saturday, 10 October 2009
var/spool/asterisk/outgoing/t.call
Does this just not work? Or am I missing something?
Thanks in advance,
--
Take care and have fun,
Mike Diehl.
___
-- Bandwidth and Colocation Provided
ave it installed somewhere in production). Sniff the
> packets coming out of it to see if you can determine its IP, but I am
> guessing if the previous owner already disabled the IVR, they probably
> locked down the device pretty well :) Are you sure it is an NA?
>
> j
>
> On M
Hi all.
I received a PAP2T-NA from a potential customer to see if I could get it
configured for testing. I plugged it into my network and plugged a phone
into it and attempted to do a factory reset from the handset.
I pressed "" and got NOTHING! Just silence. So, is this TA a brick? Or
nd on the other end of the line before
> speaking. For example, when I answer a call, I don't say, "Hello" until
> I hear a bit of noise on the channel which takes a second. If it's
> longer than about a second maybe you have some other issues to deal with.
>
>
Hi all,
I've got a problem where many times, there is silence at the first 1-2
seconds of a call. Then it clears up and it's crystal clear. I've not
put a sniffer on it, yet, but I suspect that the media channel is still
being set up. The server shouldn't be too overloaded. Can anyone give
me
On Fri, 2009-02-20 at 13:46 +1300, Michael wrote:
> > I also think you should check the economic stimulus package. There are
> > billions of dollars allocated to ISPs. It could be a windfall.
>
> Yoohoo! Let's print some more money. I don't think that's been tried
> before
Of course it has
Hi all,
I'm looking into being able to send/receive SMS messages with my
asterisk box in the US. I've seen the SMS command as well as the Kannel
program. I'd prefer to do it from Asterisk.
I've tried something like:
exten => 999,n,sms(15551234567,s,"This is a test")
in my dialplan, but when t
On Thursday 24 July 2008 12:58:29 am Steve Edwards wrote:
>
> The "agi debug" command (1.2) would have shown you where you violated the
> "protocol."
Nice to know...
--
Mike Diehl
___
-- Bandwidth and Colocati
worked as expected.
Thank you for your time.
Mike.
On Wednesday 23 July 2008 01:48:14 pm David Van Ginneken wrote:
> Mike Diehl wrote:
> > Hi all,
> >
> > I'm trying to build an IVR using the Perl AGI module at
> > http://search.cpan.org/~jamesgol/asterisk-perl-0.10
101 - 200 of 242 matches
Mail list logo
