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to do this? The complaint we are getting now is the call
rep doesn't want their phone to ring when making a call. Can the manager
interface give a phone number to dial on an off hook Zap line?
Thanks!
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Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
O
y a Cisco issue.
>
> You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs.
>
> tracinet wrote:
> > Jason,
> > I am at least having similar issues with rfc2833 DTMF:
> >
> > http://bugs.digium.com/view.php?id=10058
> >
> >
> > On 6/20/0
Two reccomendations:
1) Enable nat for the SIP channels of the phones in SIP.conf.
Or
2) If all the remote phones are in the same location, an IPSEC tunnel
between the remote router, and your Asterisk machine.
Jason.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
t be seen as a 'solution',
as callprogress has it's place (disconnection detection, etc).
Don, have any changed been made to your zapata.conf immediately before this
issue started occuring?
Jason.
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The problem was with ACPI screwing up interrupt routing. Added
pci=routeirq
to /boot/grub/grub.conf to turn off acpi for interrupt routing. Now
I've got two green LEDs.
Thanks to Jolan Luff for figuring this out!
Jason :)
>> On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wr
Yes, both cards are jumpered for E1.
Any other ideas?
Jason :)
James Texter wrote:
> Have you checked to ensure the card in server #2 is jumpered for E1?
>
> On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote:
>> Hi there,
>>
>> I've got two Asteris
RED alarm on
the other and no LED light as soon as the wct2xxp driver is loaded?
Thanks for the help,
Jason Carter
DLS Internet Services
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Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite>Asterisk--->Cisco SIP proxy>SIP Conference service
The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any dig
Hi guys,
Does anybody try to install IPV6 support on asterisk?I just found a patch
for that but it is released on 2005,I have no idea if there is new version
to support ipv6 or new patches,please advise.Thanks a lot.
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I ran the gambit and eventually, against my better judgement, I
finally broke down and installed HylaFax+IAXModem and I have had
absolutely zero problems with it. I'm extremely impressed.
This is the how-to I followed. A small amount of the instructions
aren't extremely clear; there is a
g an older 1.2 version of Asterisk.
Is anyone able to confirm the same behavior in newer versions? Is there a way
for Asterisk voicemail to behave like regular voicemail where a message remains
"New" until the caller does something to it (other than simply l
We have a job that requires extensive knowledge of asterisk queues. The
work can be done remotely. Our customer is looking to completely
overhaul their current queue structure. Please contact me offlist if
you are interested or need more details.
- Jason
, Jun 05, 2007 at 06:35:15PM -0600, Stephen Bosch wrote:
> Fuermann, Jason Bryce wrote:
> > http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
>
> This only works if you have a reseller account.
>
> -Stephen-
>
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, June 05, 2007 1:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Where to find Polyco
ked with Skinny firmware. There is
support for it in chan_skinny in 1.4, but it's mostly useless, as they can only
be line appearances. svn trunk has support for speeddials/hints though.
--
Jason Parker
Digium
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g in linux.
>
> Ideas?
>
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB
1.4 is not trunk. You need to install svn trunk
(http://svn.digium.com/svn/asterisk-addons/trunk/) if you want to use
chan_mobile.
--
Jason Parker
Digium
_
I know Citel offers a 24 port device.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Thursday, May 31, 2007 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] High Port Count ATA
I'm trying to find a high port count
Hi,
I want to transfer the call to a conferencing
room while dialing.
I tried to do that using manager API(Redirect),
but it did't work.
Regards,
Jason.
Don't pick lemons.
See all the new 2007 car
Buddies,
I am new guy here,I installed Asterisk 1.44 and setup AsteriskNow
manually.Iwant to disable the global digest authentication for
registration so that I
can easily to test my Asterisk system with another call generation tool,how
can I do that?Will appreciate for any replies.Thanks in advan
ng me think through this!
- Jason
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6:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn
I have also only tested but it did work well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Thursday, May 10, 2007 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CITEL gateway does i
ly well. There were some mic issues, but those were
driver related.
idefisk isn't open source, but there is a free version with a bunch of features.
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Jason Parker
Digium
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asterisk
I couldn't find it in Zaptel 1.2.17.1 nor 1.4.2.1 changelog.
>
> I believe it was introduced in Zaptel 1.2.17.1. From the description
> of zaptel 1.2.17.1 posted to www.asterisk.org:
>
> "Added the ability to monitor pre-echo cancellation au
Isn't that the function of an attended transfer? User3 hears User1 to
see if they want to take the call or not. User1 should then hit the
transfer key again to finalize the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber
Sent: Thursday, May 03, 2007 12:54 PM
To:
I've used these gateways and never experienced any of these problems. I
could imagine me missing the popping noise but I do know that MWI did
work just fine.
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Bo
I've had mixed results with changing ulimit and not restarting asterisk.
Best bet is to stop and start asterisk so that it calls a new shell
Rilawich Ango wrote:
Thanks for your reply.
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Bo
e for the bump on my
head to go away. :)
Anyone trying to get this to work would be better off getting a phone
that does support hinting, and has plenty of documentation on how to do
it (Polycom, Aastra, SNOM, etc).
Jason Howk wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I ru
don't know why, I just know that when my ulimit was set at 1024 I was
getting around 120 concurrent calls before getting the error.
Tzafrir Cohen wrote:
On Thu, Apr 26, 2007 at 08:43:17AM -0500, Jason Fuermann wrote:
1024 open files will get you around 120 concurrent calls.
8
1024 open files will get you around 120 concurrent calls. Rilawich,
putting the ulimit in safe asterisk doesn't always work (my experience,
and proven because your ulimit -n is still 1024). Add this line in
limits.conf "* - nofile 65535", its
located in /etc/securi
post it on the internet.
Asterisk is supposed to be more skinny friendly these days.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Howk
Sent: Wednesday, April 25, 2007 7:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
7960.
If anyone has gotten this to work, I'd like to hear about it.
--Jason.
John C. Wolosuk Jr. wrote:
> Has anyone had any success with getting SLA going between 2 SIP phones?
> (Particularly a set of Cisco 79xx's) The SLA document that comes with
> the asterisk source is
't
help there. If you want/need anything, config files, commands run, just
let me know. I'll be glad to help.
--Jason.
David Olsen wrote:
> On 2007-04-19 at 13:09:51, Doug Lytle <[EMAIL PROTECTED]> wrote:
>> Do you see anything weird when logging (telnet to the ip) into t
Running 1.4.2 with a 7960G v. 8.6 and all is well...
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will not
type for Australia (obsolete)
Jason Aarons
Consultant
http://www.dimensiondata.com/na <http://www.dimensiondata.com/na>
904-338-3245 cell
For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-97
I am looking to allow some users to login to a website and change where
their ext is forwarded to. any ideas? It can be very simple or I can
install a full package and then allow certain users certain access.
Thanks in advance
Jason
also I've seen that not having the correct version of sip.cfg and
phone1.cfg could cause weird problems. Make sure you are using the ones
that came with the firmware.
Mike wrote:
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again,
- [EMAIL PROTECTED] wrote:
> [snip]
>
> I have a feeling I'm forgetting something fairly easy and stupid, but
> I
> can't seem to see what it is. Anyone have any suggestions?
>
Dial(SCCP/[EMAIL PROTECTED])
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Digium
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e client.
Any suggestions from those who know?
Jason
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If you set the queue strategy to ringall it should ring all the
interfaces you have set up in that queue. Just make sure you have
member => SIP/EXT setup.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
Novak
Sent: Thursday, April 05, 2007 4:06 PM
To: asterisk-users@lists
e, I
can hear it on the softphone perfectly but the noise fills all the gaps
in the audio. If I talk into the softphone, I can hear it on the hard
phone but the audio is a bit soft and distorted.
I'm stumped on this. I've never ran into this type of audio problem
before.
Has anyon
Is it exists?
Regards,
Hong
Now that's room service! Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097
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nd go to another
context/entension in the 'non-realtime world'. Is this not possible? Is
it all or nothing with Realtime?
Thanks,
Jason Wolfe
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- "Chris Nighswonger" <[EMAIL PROTECTED]> wrote:
> Jason,
> Ok, the 30VIP template seems to be working ok as far as button
> assignment goes. I can define speeddial numbers to the speeddial
> buttons. However, it appears that there is no code to support the
> S
I have my system set up to check the cid of the calling number and if
the room number the user inputs matches the calling extension (the last
4 digits in my case) then the number is considered admin. This does have
the same downside that Dovid pointed out, the admin must be in the room
for user
- "Chris Nighswonger" <[EMAIL PROTECTED]> wrote:
> On 3/29/07, Jason Parker <[EMAIL PROTECTED]> wrote:
> > It should be immediately obvious how it works.
>
> Maybe to some who have been in on the skinny/cisco conversation for
> awhile. I am not new to c
- "Derek Whitten" <[EMAIL PROTECTED]> wrote:
> if i remember right, most of the buttons on those and the 12SP+ phones
> don't really work
> because there isn't a button template in *
There is a button template, the problem is that most of the softkeys sim
> etc.).
>
> Thanks,
> Chris
Search the code for "30VIP", there are only like 2-3 places where it's
referenced.
It should be immediately obvious how it works.
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ing to the user?
Thanks,
Jason Wolfe
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ople would have to subscribe to, and
many of the questions/answers for one version are quite relevant to the other.
This fracturing of the community would be very silly in my opinion, and is
extremely unlikely to happen.
--
Jason Parker
Digium
___
--
I do not have any answer int he dialplan. what I mean is that when I
call any other SIP phone is does the answer in the CLI. Even if I put
and answer() in the dialplan still no ringing
Jason
Luki wrote:
shouldn't there be an answer in there somewhere?... like...
No... you can
#x27;en')
It is registered and will make calls but I never get the
-- SIP/phil is ringing
This happening on my 2 linksys phones only
Jason
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Good Idea, but when the user has to do nothing is better for my users!
Thanks
JAson
Mojo with Horan & Company, LLC wrote:
Another option is to have the user hit the forward button on their
phone and manually type in their cellphone number when they're going
to be out of the office
exten => 111,1,Wait(1)
exten => 111,2,Playback(Randy)
exten => 111,3,Dial(Sip/Randy,20)
exten => 111,4,Goto(111-${DIALSTATUS},1)
exten => 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten => 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
Thanks a lot
Jason
Doug
I have a SIP user and a remote IAX device
I want both to ring 3 times then if neiter pick up it to go to the next
thing in the dialplan. Can you do this from the dialplan or do I need
to set a hunt group up?
Thanks
Jason
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Thanks Jason
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Answers in-line...
Hope this helps!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alan
Chandler
Sent: Wednesday, February 28, 2007 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie Planning Help
for regular stuff.
what about for playing voiceprompts?
>
> Jason, if you do a 'vmstat 1' on the unix prompt when a call is run,
> does it ever hit an idle count of 0 somewhere ? If so, you have
> performance issues, if not, you'd probably look toward the network, or
>
es, I have been looking into that after reading Steve's response.
Unfortunately I get a compile error with it. I'll try a newer kernel.
I have a pure SIP installation also
> Jason - is this on a standard PC motherboard (or a mini device like Linksys
> WRT)?
>
Yes, stand
Steve Murphy wrote:
> On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote:
>> I have been testing asterisk 1.4 with a view to deploying it in my
>> organisation and I am experiencing jittery voice prompts from the voice
>> mail system. I get this jitter even if I try a simp
dicate an issue.
I read somewhere that disabling X can help, but it did not in my case.
I am at a loss as to how I might track down the problem and fix it. Any
pointers would be greatly appreciated.
Thanks,
Jason
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Glad to hear you had a workaround.
I would suggest re-queing your TAC case, perhaps you got a outsourced or less
experienced engineer at Cisco. Their support has varied depending on which
city/group you get. Some have more experience then others.
While your 2600 from 2001 timeframe it should
1.2.1
Jason Wolfe, CTO
Click For A Call, Inc.
[EMAIL PROTECTED]
1-800-218-4951
o (770) 287-0273
c (770) 561-6956
This e-mail transmission may contain information that is proprietary,
privileged and/or confidential and is intended exclusively for the person(s) to
whom it is addressed. Any use
was bridged, but the call was still continuing.
Any thoughts on where to start debugging?
Jason
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about this?
Jason
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Would you attach your whole zaptel.conf and
zapata.conf?
--- C F <[EMAIL PROTECTED]> wrote:
> Also check out immediate=no
>
> On 2/18/07, Eric ManxPower Wieling <[EMAIL PROTECTED]>
> wrote:
> > Eric "ManxPower" Wieling wrote:
> > > David Ruggles wrote:
> > >> I'm sending 12345 as DNIS on a Wink
we have this problem. In our case it was due to the voice mail app; it
was failing to unlink files in memory when creating mp3s. Not sure what
your specific problem might be
Giorgio Incantalupo wrote:
Hi,
my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I
found the following
Your best bet is to use DUNDI
Azfhasterisk wrote:
Can someone point me to some documentation on how to configure an
Asterisk box to do Termination and Origination for a few other
Asterisk servers? We have a box with a T-1 in it and we want to share
it with some other companies that have Aste
Hello,
I have the following simple application...
1. Call is answered, and Dial() function is used with a macro to dial
out to a number.
2. 'Called' party answers the phone, and hears a message (this is a
function of the macro)
At this point I'd like for the 'Called' Party to be able to mak
set up (if its
software then that may be causing it)
From: [EMAIL PROTECTED] on behalf of Yuan LIU
Sent: Thu 2/8/2007 1:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and 802.11g
>From: Jason Fuermann <[EMAIL PROTECTED]&
I have a telco providing DTMF inband, they say they can't provide it any
other way. This is creating headaches for me.
What is the common method for SIP DTMF? Kpml, or 2833 or inband?
My handsets don't support inband so I'm tying up some expensive
resources to convert the inband DTMF to ou
your asterisk box has to do audio conversion, its getting bogged down
Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.
Here's the topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
our Polycoms reregister almost immediately. I think the problem your
running into is that when the softphone is registered the polycom gets
some kind of error from asterisk which prevents it from reregistering
Rob Schall wrote:
That's what I would have thought. I set the timeout to be 300 secs
We have a similar situation and we do a realtime lookup in an external
db, works like a champ
Steve Murphy wrote:
On Wed, 2007-02-07 at 22:21 -0500, Lee Jenkins wrote:
Eric Germann wrote:
We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint
PCS and one for Alltel.
Hi,
This is the configuration I want.
Hard Video phone<---video--->Soft Video Phone(PC)
^
|
audio
|
V
Audio Only Phone
Any idea?
Regards,
Jason
Do you Yahoo!?
Ev
We have done limited testing with the Citel gateways and they seem
pretty cool. We're fixing to deploy them as a replacement to a hotel
pbx, and after that use them as an interim solution until full VoIP
convergence in our campus environment. I would be interested to know
what other peoples exp
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.cfg file look
like?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue
From what I know this log show
Fixed that issue but it does not change the error
0126204105|cfg |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr
1 of 1)
0126204105|cfg |3|00|Downloaded application image is identical to
current version
0126204105|cfg |3|00|Phone suc
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,
it errors out with a 0x1 error
Any Ideas?
1005195711|so |4|00|-- Initial log entry --
1005195711|so |4|00|+++ Note that bootrom log time
Its a problem in your database. something might have corrupted...be
prepared to load a backup
Gregory Duchatelet wrote:
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and
today I encountered this error.
Now, I have no acces to any information in mysql realtime
Yes it should, I'm not running bleeding edge 1.2 but it isn't an older
branch either.
Mark Johnson wrote:
Jason Fuermann wrote:
I'm not sure about the sippeer stuff, or where they get that
variable. We lookup our info in a database to set it. Also to use
sipcalledrpid you
I'm not sure about the sippeer stuff, or where they get that variable.
We lookup our info in a database to set it. Also to use sipcalledrpid
you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 .
Rob Schall wrote:
Here is what I have in my extensions.conf file now. Trustrc
try actually setting the rpid in the dialplan using
sipcalledrpid(name,number)
Rob Schall wrote:
I set both the trustrpid and sendrpid to "yes", but the calling phone
still doesn't show the caller id of the person they are calling.
Jason Fuermann wrote:
check out rpid
Mar
check out rpid
Mark Johnson wrote:
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does i
gt; -A.
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__
=Ring Answer)
>
> exten =>
> PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
>
> exten => PAGE2105,n,Dial(SIP/2105,5)
>
> exten => PAGE2105,n(skipself),Noop(Not paging
> originator)
>
>
>
>
>
> exten => Debug,1,Noop(dialstr is
>
LOCA
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten => _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten => _99,n,SIPAddHeader(Call-Info:
\;answer-after=0)
;exten => _99X
gt;
> asterisk-users mailing list
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Jason Parker
Digium
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to us something that
requires a timing source, such as meetme or iax trunking."
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Jason Parker
Digium
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ke a small
> project to bother with SF.net.
> --
>
> Warm Regards,
>
> Lee
>
Why not just post the text of the AGI to the wiki page?
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Jason Parker
Digium
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asterisk-u
Hi,
I configured openh323_v1_18_0, pwlib_v1_10_0 and
asterisk-oh323-0.7.3.
I can call inbound and outbound.
But early media is not working in outboubd.
Regards,
Jason.
oh323.conf
==
[general]
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
007"
Best Regards
Josué
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- John French <[EMAIL PROTECTED]> wrote:
> I have CentOS 4.4 x86_64 running on an Pentium D 830 dual core
> processor
> with the smp kernel. Does Asterisk need to be compiled in any special
>
> way to gain performance benefits from this setup?
nope
--
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David
> Thomas
> Sent: Friday, December 29, 2006 8:18 AM
> To: Asterisk Users Mailing List - Non-Commercial Discus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Friday, December 29, 2006 8:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 Random disconnects
On 12/28/06, Jason Adams <[EM
. It seems like asterisk gets hung up on a certain call and dumps.
Anyone else noticing anything like this?
Thanks,
Jason
Jason Adams
Sumo Systems
4694 Cemetery Road
Suite 310
Hilliard, OH 43026
Phone | 614.433.9906 ext: 102
Fax | 614.433.9931
E-mail | [EMAIL PROTECTED] mailto:[EMAIL PROT
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