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
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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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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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 Asterisk hosted PBX servers with Digium TE210P cards
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 wrote:
Hi
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-liteAsterisk---Cisco SIP proxySIP Conference service
The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any
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
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-
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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
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 listening to it)
?
Thanks.
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need to install svn trunk
(http://svn.digium.com/svn/asterisk-addons/trunk/) if you want to use
chan_mobile.
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has support for speeddials/hints though.
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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
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
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 cars at Yahoo
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
this!
- Jason
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complains:
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
I haven't found any other information regarding these errors. I am just
wondering if they are bugs. Any insight would be appreciated!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road
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
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 audio with
ztmonitor
- Noah
Yes, you are correct.
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with a bunch of features.
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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
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
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
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
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 file
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 run the phone with sip
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 about as clear as mud.
Does anyone have a working
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: Re: [asterisk-users
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 the phone
and doing a show register?
I
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
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 800-974-6584
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
- [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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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:
suggestions from those who know?
Jason
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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 anyone seen this before and found a solution?
Below is Zapata.conf and Zaptel.conf
Thanks!
Jason
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
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
- 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
STIMULUS_SPEEDDIAL case. Is this correct
/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:
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 or c++, but am to * and cisco ip phones
, there are only like 2-3 places where it's
referenced.
It should be immediately obvious how it works.
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- 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 simply
aren't implemented.
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to the user?
Thanks,
Jason Wolfe
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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.
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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
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.
Jason
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 Lytle wrote:
Mike wrote
?
Thanks Jason
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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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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
snip
.
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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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 simple hello world dial
plan
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, standard PC (although older as mentioned in previous post)
Thanks,
Jason
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
perhaps a silly Voice Activation setting in your phone.
I
, but the call was still continuing.
Any thoughts on where to start debugging?
Jason
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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
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
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 Start T1.
In case
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
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
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
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]
Date: Thu
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
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
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
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
|
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!?
Everyone
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
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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
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
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 everything working Perfect.
Then it goes to the Welcome screen then after a long 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
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
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
Mark Johnson 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'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.
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'll probably need
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PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi
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:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n
-users
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a timing source, such as meetme or iax trunking.
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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
with SF.net.
--
Warm Regards,
Lee
Why not just post the text of the AGI to the wiki page?
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Best Regards
Josué
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-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 [EMAIL PROTECTED
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 Discussion
Subject: Re: [asterisk
- 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
--
Jason Parker
Digium
. 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] blocked::mailto:[EMAIL
to the list and would like to know how to search it so that
I do not post any questions that have already been answered (like this one)
- Mark
--
Jason Parker
Digium
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Jason Parker
Digium
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-2.6.16.13-4-obj/i386/smp
make[1]: *** [linux26] Error 2
make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2
make: *** [all] Error 2
Thanks
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Jason Parker
Digium
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