are secrets too much hassle? You set the password once and forget
it. With the Aastra phones you could setup phone provisioning files to
automate the process.
Ryan
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data. I process this
data and insert it into the companies call database link to users, you
could just email it. I basically added a column to mysql and mark each
row as processed.
Ryan
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with Exchange UM. I've
found 1.6.1.18 to work all around with only a minor DTMF issue with
Exchange UM that I was able to patch.
Ryan
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New to Asterisk? Join us
.
Thanks,
Ryan
DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute-
INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0
...
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X
I wasn't sure how the lines were counted. Here is the debug output
from Asterisk where it is building the invite packet. I looked at the
a=T38 lines and nothing is standing out to me.
Ryan
[May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 0 [ 47]: INVITE
sip:+num...@x.x.x.x:5060 SIP/2.0
[May
On Thu, May 6, 2010 at 7:11 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3
the calls I can easily transition users by
setting them up a SIP hardphone and then removing the follow me.
Eventually as funds allow I can move everyone over to a SIP phone.
Then it is as simply as turning the Toshiba off.
Ryan
On Mon, May 3, 2010 at 10:30 AM, Eddie Mikell ed...@rimmkaufman.com
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
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Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
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Check out the 'p' option for the Dial command.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
It enables call screening, so you have to press 1 to answer. This can also
prevent the voice mail from being left on your cell phone.
--
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
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Catches 555 through 559:
exten = _55[5-9],1,answer
exten = _55[5-9],n,playback(beep)
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
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Ah, sorry, I totally missed that in your description.
Other than the speech recognition that Danny is suggesting, my only thought
is to use an agi that will originate another leg, run AMD (answering machine
detect) and then dump the two parties into a conference to re-join them(or
use the Bridge
Are you running asterisk in a virtual machine?
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So I be it sounds like all the recordings are underwater.
Are you using dahdi for timing? Can you run dahdi_test?
Asterisk needs a good timing source, in the case when you don't have a
physical card providing it, it relies on kernel ticks or the RTC (or HPET).
Because of the nature of virtual
Have you tried 'type = friend', might also want to make sure 'allowguest' is
set to 'no', as this may be putting guest calls into your default context.
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1.6.1.18, DAHDI
2.2.1, libPRI 1.4.10.2, and Sangoma wanpipe driver 3.5.6. I had some
issues with newer wanpipe drivers and kernel soft locks. I also had a
PCI dma timeout issue which required a Sangoma firmware update. Since
then it has been rock solid since with 22 days of uptime.
Ryan
On Wed, Mar 24
Hey Philipp,
You can check out
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk for
setting up from brute force detection and blocking with asterisk. There are
also a link at the bottom about rate limiting registrations via iptables.
--
Please post your results as a note for the issue.
Thanks.
Ryan Bullock
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Hello,
I am having a problem with my current SIP over VPN setup.
We have a server running asterisk at our office. All the phones in the
office are on the same network / local to this server. We also have two
employees with home offices using SIP phones over VPN to connect to the
asterisk
canreinvite=no did the trick! Thanks!!
[Cary Fitch]
One thought: if you are using reinvite try turning that off. That will be
a clue.
It would seem that both phones are on the local net via VPN, and should be
able to talk to each other if they can talk to anyone in the office. (As you
The VPN termination is a Cisco/Linksys RV042.
We have that solution running fine...
Is your VPN termination a Linux box? Is it also the office router? Is it
also the firewall?
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to replace an older Asterisk server within the next
month,
Ryan
On Wed, Dec 23, 2009 at 4:48 PM, Fred Posner f...@teamforrest.com wrote:
On Dec 23, 2009, at 4:21 PM, Sascha Ferley wrote:
Hi,
I am in need of ordering a new server here for our asterisk solution. Since
the corporate standard is Dell we
looking at the extension list.
Ryan
On Wed, Nov 25, 2009 at 11:58 AM, tom tomabr...@gmail.com wrote:
hi,
we are running a switchvox system, and i would like to know what the
practice is for users who are working party in the main office and on some
other days with their laptops either from home
is seamless as they can 4 digit dial any extension and
the call will be routed to the correct system. This does take a bit of
duplicate setup on the two systems, but was worth the hassle for the
end result.
Ryan
On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov
abalas...@evaristesys.com wrote:
PRI
I agree with the cross talk analysis. My suggestion would be to focus your
efforts on the analog trunks/stations, not SIP. Are you using twisted pair or
shielded cables for your analog runs? If not, you might consider changing the
cables or at least increasing the physical distance between
using the php imap_open command,
which is also built against the c-client libraries, though they appear
to be built against the older 2001 version.
Thanks,
Ryan Wyse
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copies the files to the
install point. Running make install on a different version will just
overwrite all installed files with those files.
Ryan
On Mon, Oct 5, 2009 at 12:53 PM, Bart Fisher b...@icpage.com wrote:
I currently have asterisk-1.4.26.2 installed and working. It was sugguested
I try
-c876bf18,
AMPUSERCIDNAME=Ryan Wagoner) in new stack
-- Executing [...@macro-user-callerid:6] GotoIf(SIP/8678-c876bf18,
0?report) in new stack
-- Executing [...@macro-user-callerid:7] Set(SIP/8678-c876bf18,
AMPUSERCID=8678) in new stack
-- Executing [...@macro-user-callerid:8] Set(SIP/8678
to queue these digits so DAHDI can send them in
response to the SETUP ACKNOWLEDGE message. What should be happening is
Asterisk sends 9 via the SETUP message, waits for the SETUP
ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message.
Ryan
Is there a way in the dialplan to figure out which agent in a ring all queue
answered a line? I'd like to take specific action based on the agent upon
hangup.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando
have spare to play
with this tomorrow instead of using my live system, with the newest
libraries, but maybe this is a signaling issue I'm just not understanding,
something this card can't even do, or something else that I've completely
missed. Any ideas?
Cheers,
-Ryan
Does anybody know if idle-url works for Cisco 79xx using Sip? If it doesn't
work is it a Sip vs SCCP issue or Asterisk vs CallManager issue? Thanks
Paul
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Has anyone successfully gotten a HiPath system to route calls over to a * box?
If so, I'd appreciate a quick consult. I've configured the HG card to look for
the * server but it doesn't seem to actually be connecting.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt
Any 2000+ user Asterisk PBX installs out there?
Please hit me off-list, I need some support on a 2000+ user Asterisk PBX
with high availability and over 10E1s to PTOs
Femi
I would be interested in some of the replies if you wanted to continue the
topic on-list... Your problem might help
= s,6,Wait(10)
exten = i,1,Goto(TT_VO,s,1)
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120 - Main Office
(407) 841-9726 - Fax
http://www.rissman.com
it
should help you figure out how much bandwidth you will need.
Ryan
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sure that an Asterisk developer can chime in and give several examples
of how Asterisk uses its threads to increase scalability. That said, there
will be a point where the number of core/CPU's won't be the bottleneck so
adding more won't help anything.
Ryan
At 11:53 AM 1/18/2008, you wrote:
Apart from the fact asterisk 1.2 is in security maintenance
mode only and wont get any other bugfixes it will be ok.
Please consider using 1.4 as it's the official latest stable
version.
Although for some of us, or at least me, no version of 1.4 has run
for
and would help make a significant impact on this family.
If you have a DID available, please let me know. I don't mind paying for the
service but I'm having difficulty quickly in locating a provider.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A
We have an issue with Linksys SPA2102-NA ATA's where there is a several second
delay between when you finish dialing and when it sends the commands on to *.
Has anyone else seen this before? If so, is there a quick/easy solution?
Ryan M. Colbert
Director of Information Technology
Rissman
Tilghman Lesher wrote:
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote:
Darryl Dunkin wrote:
You can store most of the configurations in a database which may be
more
accessable to you.
Perl can also parse these configurations quickly enough if you know
how
to use the input
PBX...
-Philip
Philip,
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
Ryan
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In article
[EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500
to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 -
((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard round to nearest integer
on 2007-07-11 00:21:57 UTC
Uname -a: Linux XX 2.6.9-55.0.2.EL #1 Tue Jun 26 14:08:18 EDT 2007 i686
athlon i386 GNU/Linux
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct
: Unable to find a
codec translation path from ulaw to unknown
[Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to
set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown'
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201
Interesting. Is the upgrade difficult? I've not attempt to upgrade our
production environment yet.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120
is listening on the correct ports, and receiving the traffic, as no
ICMP messages are being generated to say that the packets could not be
delivered.
Ryan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 5:38 PM
To: Asterisk Users Mailing
traffic flow that I see is as
follows (one way traffic into Asterisk)
SIP Phone --- Media Gateway --- Asterisk --- SIP Phone
Ryan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava
Sent: Sunday, 11 November 2007 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi Vivek,
I'm not sure what you mean, could you explain further?
Regards
Ryan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 1:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP
ends.
It doesn't seem to be complaining or generating any errors that I can see, any
suggestions on what I can do or where to look to find out what is going on?
Thanks in advance,
Ryan
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server doesn't do
anything with the packets it gets from the near-end SIP phone and the media
gateway.
SIP Phone - Media Gateway - Asterisk - SIP Phone
An asterisk internal call will work fine. Eg;
SIP Phone - Asterisk - SIP Phone
Regards
Ryan
-Original Message-
From: [EMAIL PROTECTED
David Gomillion wrote:
Ryan Stille wrote:
Don Pobanz wrote:
Ryan Stille wrote:
I have a ring group setup that I'd like to ring a bunch of local
extensions, plus a few outside lines. I want recipients to
confirm the call by pressing 1 before
Don Pobanz wrote:
Ryan Stille wrote:
I have a ring group setup that I'd like to ring a bunch of local
extensions, plus a few outside lines. I want recipients to
confirm the call by pressing 1 before they are connected.
But when I add in an external number to this ring group
ring any more.
Any ideas?
Thanks,
-Ryan
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I tried to use it back in 1.4.6 or so and it is horribly broken, I ended up
rewriting the functionality with dynamic queue members in the dial plan. I
really liked the call back agent feature set. I found it to be far superior
to dynamic queue member alternative.
-Ryan
On 9/12/07, Anthony
: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Contact: 6000 sip:[EMAIL PROTECTED];transport=udp
-
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK
***debugging output*
thanks in advanced
Ryan
I'd be interesting in pooling resources for this. We've seen the success of
Vonage's Visual Voicemail and would like to emulate a similar solution.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
I've had requests to processes incoming voicemails with voice recognition
routine and add the output text to the body of the email message from * with
the attached .wav file. Has anyone implemented this type of feature and
willing to share some notes?
Thanks!
Ryan M. Colbert
Director
The 7914 only works under SCCP; the SIP firmware does not support it at
all (the expansion panel won't even power on fully.) The SCCP channel
driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been
much discussion
-- Average: -21.255493
Any help would be appreciated!
Thanks,
Ryan
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I've had decent luck with PhonerLite, connecting via SIP. The interface
is not the best, but I've been able to connect reliably and make calls.
-Ryan
bilal ghayyad wrote:
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one
If anyone can help me out on this and is interesting in seeing my kernel
config or something else please let me know.
Thanks,
Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew J.
Roth
Sent: Tuesday, July 24, 2007 12:47 PM
To: Asterisk Users
.
Thanks,
-Ryan
David Gomillion wrote:
On 7/19/07, *Ryan Stille* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Right now I've been working on setting up an Trixbox server on our
internal network. Its behind the firewall, but I'd like to open
up the
firewall to it because
. Is that right?
Right now thats ports, I've read that you can chop that down to 20
ports for just a few calls. We want to have 5-6 simultaneous calls, so
if I set rtpstart to 10001 and rtpend to 10100, then open up those
ports, is that adequate?
Thanks for any help.
-Ryan
Also the, the firewall does NAT for the server, it sounds like this may
cause some issues for my SIP clients?
-Ryan
Ryan Stille wrote:
Right now I've been working on setting up an Trixbox server on our
internal network. Its behind the firewall, but I'd like to open up the
firewall
,
Mike Ryan
Installation Support Engineer
Percipia, Inc.
858 Morrison Rd.
Gahanna, OH 43230
+1 614-856-1123 (office)
+1 614-579-6055 (cell)
+1 614-751-2018 (fax)
mykryen (skype)
[EMAIL PROTECTED] (yahoo)
[EMAIL PROTECTED] (msn)
[EMAIL PROTECTED] (gtalk)
http://www.percipia.com
it? Or could he just log on twice with the same
extension/secret?
Thanks,
-Ryan
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externally.
Something else I've thought about was playing a message to the user
asking them if they are sure they want to connect, and if so press 1 -
then I'll forward out the call.
-Ryan
Gustavo Hernandez wrote:
use folow-me
- Original Message -
From: Ryan Stille [EMAIL PROTECTED
= 101,1,Dial(SIP/lesnet/12185551212,30,tpm)
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Hope that helps.
Ryan
. Below is the output of asterisk - -c
-U asterisk, and the relevant output of show dialplan.
Note that the sip calls come in on extension 666.
Thanks much in advance.
Ryan
===
show dialplan:
[ Context 'in-from-7869101
Ryan Goldberg wrote:
So, incoming calls on zap work just as I expect them - an intro is played,
the
Ah, ignore all that- it had to do with caller id being empty vs unknown or
something of that nature - at any rate some problem I can solve myself. I
jumped the gun by posting.
Ryan
menu in
Trixbox? (where I'll ask users to dial the party's extension or press 1
for sales, etc.) I've dug around in the menus but I don't see anything
resembling this.
Thanks,
-Ryan
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for service. Do you think the hardware is adequate? If
there's a chance its not enough horsepower I want to find a different
server.
Thanks much,
-Ryan
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asterisk-users mailing list
: Monday, February 19, 2007 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Problem with busydetect and cell phones
Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet
Original Message-
Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet for an answer to this one. My phone
company
provides no disconnect supervision. Hence I'm forced to use the
busydetect
feature. I have
The landlines are with ATT. The cell phones I'm testing with are Cingular
(ATT subsidiary).
On Fri, Feb 16, 2007 at 11:47:02PM -0800, Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered already.
I've scoured the internet for an answer to this one. My phone
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet for an answer to this one. My phone company
provides no disconnect supervision. Hence I'm forced to use the
busydetect
feature. I have a TDM400 with two FXO ports. If I call from an
internal
I have a very strange problem I'm hoping someone has encountered already.
I've scoured the internet for an answer to this one. My phone company
provides no disconnect supervision. Hence I'm forced to use the busydetect
feature. I have a TDM400 with two FXO ports. If I call from an internal
? I've searched this list back through June 2005 and I don't see
anything that helps and I've spent hours searching on google only ending up
with dead ends. Any pointers would be greatly appreciated.
Thanks,
-Ryan
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Asterisk does not work very well in a VM
due to the timeslicing. Dropped calls, jittery audio and echo can all creep in.
Good news is that an AD controller runs
just fine in VMware. Just make sure the box has enough RAM to keep it happy,
and use a physical second disk for the Windows
I spent quite a bit of time debugging the
7935/7936, and it is an issue inside the firmware that Cisco knows how to work
around in CallManager. There are better conference phone options available, and
development on chan_sccp is basically dead at this point anyway, so I dont
see this one
It is almost always better to use a single T1/E1 card when possible to
avoid conflicts. A Digium TE2XXP series card sounds like what you would
need. The price is usually less than buying 2 single cards.
The server itself is fine. It has 2 PCI slots, so if you went with a
single card you would be
and maybe 1k users registered before things start acting flaky. I really
appreciate the info. I'm looking forward to hearing about your current
project when you get a chance to write it up.
Thanks again,
Ryan
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Here's the relevent section of my extensions.conf:
### Handle voicemail
exten = _1XX,1,SayDigits(${EXTEN})
exten = _1XX,2,MailboxExists(${EXTEN})
exten = _1XX,3,Playback(vm-nobox)
exten = _1XX,4,Goto(teliax,5013584196,3)
exten = _1XX,103,VoiceMail(b${EXTEN})
exten =
I'm getting intermittent errors using Teliax service.
Here's an example of it:
-
[EMAIL PROTECTED] ~]# asterisk -r
Asterisk SVN-trunk-r39206, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show
The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
IMO.
I believe the phone that doesn't work quite right with the 8.2 SIP image
is the 7970. I have probably 20 7940s/7960s, all running the 8.2 SIP
image
to have to support,
I would (and do) use SIP.
-Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrea
Frigo
Sent: Thursday, June 29, 2006 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Cisco 7905G SIP firmware
Chan_sccp does not support blind transfer. I would suggest using
chan_sip and the SIP images with these phones; it is much more stable,
has more features and is being actively developed. Chan_sip supports
blind transfer and 3-way calling, plus it handles multiple calls on hold
a bit more
So I've got a 601 (1.6.6) with the side car, and the buddy watch seems to be working but it updates the statuses unreliably. When I do a sip show subscriptions in asterisk it lists my phone 12 times and at the bottom it says 0 active SIP subscriptions(s) I've got an older CVS-HEAD build, pre
1.2,
with
Asterisk with a X100P clone FXO port. I can't remember all I did before, but
I'll be sure to post my experiences in the coming weeks.
Ryan
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Polycom with different software, plus you can get
support for the Polycom through a reseller for a lot less than with Cisco.
-Ryan
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, June 15, 2006
10:19 AM
To:
asterisk-users
,
-Ryan
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is pretty ridiculous. Maybe there is a more sane vendor I should be buying from?
Thanks,-Ryan
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=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
pickupgroup=1
immediate=no
callprogress=no
relaxdtmf=yes
signalling=pri_cpe
switchtype=national
context=default
rxgain=15.0
group=1
channel = 1-23
Has anyone else had
and solved this problem?
-Ryan
. There are a myriad of applications you can use
Asterisk for, and almost no 2 users will have the exact same setup. The
asterisk config files are actually really well thought out and very easy
to parse/rewrite with small perl scripts.
-Ryan
-Original Message-
From: [EMAIL PROTECTED
that. :)
-Ryan
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I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going on.
I can run
Sounds great.
I'm just a home user of Asterisk, but I love the
product and have recommended it to alot of other people. Let us know when the
site is up.
Ryan
- Original Message -
From:
Bruce Reeves
To: asterisk-users@lists.digium.com
Sent: Thursday, April 13
I'm interested but I'm in the Dallas area. Are
there any in the Dallas area anyone knows of?
Ryan
- Original Message -
From:
Bruce Reeves
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 12:51
PM
Subject: [Asterisk-Users
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