Hi,
I am using asterisk-1.4.0.
I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop) does and
what Native bridge (bridge_native_loop) does.
I have configured my dial plans and options such that I can enter
bridge_p2p_loop. However, I am unable to enter bridge_native_loop for some
8 mar 2007 kl. 14.36 skrev René Enskat:
hello all,
My problem if i have my extensions and sipusers in a realtime
database it is not possible to use BLF or hinting.
i see only idle or unavailable status but if the phone is ringing
or in use i can't see it.
Is there a fix or any workaround?
8 mar 2007 kl. 21.05 skrev Daryl Jurbala:
OK...that makes much more sense. So here's my follow-up question:
what's the easiest way to check if I'm native bridging a call. I'm
trying to offload as much RTP traffic as possible, and want to have
a way to check quickly (there are well over
9 mar 2007 kl. 08.52 skrev Santosh Raghuram:
Hi,
I am using asterisk-1.4.0.
I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop)
does and what Native bridge (bridge_native_loop) does.
I have configured my dial plans and options such that I can enter
bridge_p2p_loop. However,
Further to this, I believe that my problem is that I'm also now running
udev.
When I compiled and installed Zaptel, I did the make install-udev step,
however the permissions in my udev directory don't look correct.
I am running Asterisk as root (this is on a debian system btw), but this
is
On Fri, Mar 09, 2007 at 04:13:04PM +0900, Mark Davies wrote:
Hi guys,
I'm hoping I've made a silly mistake here, but I've been staring at the
screen for the past few hours and I can't work it out.
I upgraded to 1.2.16 recently, and am having problems with zaptel.
The card
On Fri, Mar 09, 2007 at 06:06:01PM +0900, Mark Davies wrote:
Further to this, I believe that my problem is that I'm also now running
udev.
When I compiled and installed Zaptel, I did the make install-udev step,
however the permissions in my udev directory don't look correct.
I
On Fri, 9 Mar 2007, Zeeshan Zakaria wrote:
Hi everybody,
What is a proper setup for a medium size business with about 20 IP phones
and 20 computers. Right now they are using a regular Linksys router which we
use at homes. Their switch is also a very standard switch. Now they need to
put there
On Thu, Mar 08, 2007 at 05:01:15PM -0800, Jose Bertuzzi wrote:
Hello Everyone, I checked with zttool that sometimes after the machine boots
the order of the boards is changed like this:
â Alarms Span
Hi guys
This is my Ist post on this group. Is there any variable like ($VM_CALLERID
for voicemail mailbox) for accessing Asterisk Voicemail password which is
set through comedian mail.??
plz reply me as soon as possible
htmldivPRE class=quoteIMG height=2
Hi All,
Thanks for every one who helped me on this regard. I think i was able to
rictify the problem.
what i did is remove
callprogress=yes
usecallinpres=yes
and restart asterisk. Today i didn't report any drop calls.
Many thanks for Eric. :)
I hope this situation will continue.
Regards,
From: Drew Gibson
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
Sarge) and the behaviour of our Call Centre queues has changed
slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced as expected (estimated
Trevor,
I also have the same problem as Drew, and that isn't how mine works.
Even though I told it to announce the time, I get the first in line as
well as second in line. I've tested it up to 5 people sitting in the
queue line, and each gets the same message (space, not time).
Rob
Trevor G.
Fix on an agent? What do you mean? We make call center software and our clients
usually have 200 to more than a thousand agents, and some agents are even
working from a remote location like their homes. We have a small application
for the supervisor throught which he/she can view the status of
Hi
Has anyone tried to reproduce the following behavior that a standard phone
line does with 911.
Normally if someone calls 911 and hangs up after the call has been
established then the line is not dropped because it is held by the 911 agent.
If you pickup your phone you should still be
What kind of hardware are you using in your setup?
I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and
the parts are easily interchangeable
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
Two things.
1) This is a bug(feature) of standard analog switchs which only clear the talk
path when both sides of the call are terminated.
2) You should post this in the asterisk development list.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Patrick Fortin
Sent: Fri
Steve Prior wrote:
I read this story and thought of Allison's prompt to try not to think
about blue eyed polar bears.
Will she be banned from foreign travel now?
Steve Prior
-- snip --
WASHINGTON (Reuters) - Polar bears, sea ice and global warming are
taboo subjects, at least in public, for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If you're using all Zaptel channels for the call, it sounds like you
want operator services mode (Dial command flag).
O([x]) - Operator Services mode (Zaptel channel to Zaptel channel
only, if specified on non-Zaptel interface, it
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
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To
telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI
plugged in.
application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium
te410p (timing only, all calls over IAX)
database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB
No failures in over 2 years.
On Fri, 9
On 3/9/07, Wai Wu [EMAIL PROTECTED] wrote:
Two things.
1) This is a bug(feature) of standard analog switchs which only clear the talk
path when both sides of the call are terminated.
Well, not exactly. The call will not terminated until the caller (not
both) hangs up. I don't knew the
Mark Davies wrote:
Hi guys,
I'm hoping I've made a silly mistake here, but I've been staring at
the screen for the past few hours and I can't work it out.
I upgraded to 1.2.16 recently, and am having problems with zaptel.
The card is detected, I get a reasonable output from ztcfg
That's cool, but I doubt my systems could handle that same load ;)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
I got the same thing on a Ubuntu Dapper.
-Original Message-
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 9 Mar 2007 12:12:45 +0200
Delivered: Fri, 09 Mar 2007 06:45:09
Subject:[asterisk-users] Boot order of 2 TE110P and 1 TDM400P
Trevor G. Hammonds wrote:
From: Drew Gibson
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
Sarge) and the behaviour of our Call Centre queues has changed
slightly.
Before the upgrade, when a caller was waiting in the queue, the
estimated hold time was announced
Dears
my Internet Provider , prevents , sip connections,
between sip client(sip phone) and sip server,
(asterisk + ser) .
both of client and server are mine.
is there any solution for tunneling the sip packets?
best
Mani
All -
Next step here would probably be to open a bug on bugs.digium.com
with a full VERBOSE/DEBUG log along with associated config files so we
can troubleshoot this and fix it if there's a problem.
Thanks.
On 3/9/07, Drew Gibson [EMAIL PROTECTED] wrote:
Trevor G. Hammonds wrote:
From: Drew
Hi -
I am going to open port 5038 on my firewall so that I can use YAACID
to spawn browser popups on an incoming call. My question is, under
manager.conf, what are the suggested settings so that I can get the
browser popups only? I'll be at different IPs so I can't lock it
down that
On Fri, Mar 09, 2007 at 12:18:00PM -0300, Melcon Moraes wrote:
I got the same thing on a Ubuntu Dapper.
On Ubuntu and Debian, put your modules in the desired order in
/etc/modules .
And just in case you need to unload the module and load them again, the
asterisk init.d script in the Debian
try changing bindport of asterisk from 5060 to something else .
On 09/03/07, Pezhman Lali [EMAIL PROTECTED] wrote:
Dears
my Internet Provider , prevents , sip connections,
between sip client(sip phone) and sip server,
(asterisk + ser) .
both of client and server are mine.
is there any
Anyway, back to your question, how about your head system running an AGI
that connects to the manager interface on the IVR boxes to find out how
many calls each is currently processing? You could set a channel variable
with the least busy host name and use that in your dial statement.
Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14?
They work fine with Asterisk; most likely it's your wireless link
that's the cause of your problem. The jitter buffer will only affect
received audio, i.e. on your side, and since that is fine, you
probably don't
This probably came up before, but I have a faxing question for everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
Never mind I found it shortly after sending this :S
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles
Sent: Friday, March
Fedora Core 6
regards, Pablo.
On Thu, Mar 08, 2007 at 05:01:15PM -0800, Jose Bertuzzi wrote:
Hello Everyone, I checked with zttool that sometimes after the machine boots
the order of the boards is changed like this:
â Alarms Span
Wai Wu wrote:
Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load?
You're more courageous than I am.
-Stephen-
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[EMAIL PROTECTED] wrote:
Dear Asterisk Users Mailing List - Non-Commercial Discussion,
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
Somebody punt this jerk.
-Stephen-
___
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Anselm Martin Hoffmeister wrote:
Am Dienstag, den 06.03.2007, 05:18 -0400 schrieb Chris Mason (Lists):
Of course, it would be highly unlikely anyone on the list would want
to report Rehan...but in case anyone does:
I have been told that unsolicited commercial e-mail (I do not imply that
In my (limited) experience with rxfax, it has issues with large faxes. I
soon gave up on rxfax and switched to hylafax (which works much better).
Check the wiki for installation instructions. (And hylafax will
correctly hangup when the fax has completed/failed/whatever.)
Wes Baehr
-Original
With hw echo cancellation you are pretty much guaranteed to not have any
problems. At least with Sangoma cards. I cannot speak for the other
manufacturers. I believe most of not all HWEC also does other things to
help clean up the sound and maybe even add background noise etc. so the over
all
I didn't know you are courageous. I upgraded to 1.4 last night.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Friday, March 09, 2007 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
BTW. We only use Asterisk for a few functions. Everything else is done
on an extenal application controlling Asterisk through AMI.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Friday, March 09, 2007 12:22 PM
To: Asterisk Users Mailing
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the calls from
the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
The * boxes run AGI scripts to process the IVR calls. Can I load balance the
Hi,
Which Hylafax client do you use.
I'm after something cheap, you could use from Windows XP, as a virtual
printer and that could retrieve fax numbers from an existing directory
(Windows Address Book or Outlook or LDAP).
Regards
___
--Bandwidth and
Olivier,
For a list of your many options, see:
http://www.hylafax.org/content/Desktop_Client_Software
I'm partial to HylaFSP, but we sell it so can hardly be considered objective.
;-)
-Darren
- Original Message -
From: Olivier
To: Asterisk Users Mailing List - Non-Commercial
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote:
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
I use it on Linux and it does.
-HJC
___
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -
We're not running echo cancelling cards here. We may have 1 or 2
phone calls a month with echo, and it's primarily calls to a certain
number. When asked about the echo, I explained the difference in
price, and for the price difference, we can deal with the echos.
For the most part, for us,
I'm reading voip-info.org
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
Sorry if this is a dumb question, but:
It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want
mysql on my asterisk box I want to connect to a remote mysql server. Can I
use mysqlclient and
Philipp Kempgen wrote:
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Seems like
#include test.ael
works but
#include test.conf
does not.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
(Technically: It is not part of the ael syntax. #include and #exec are
preprocessing done before the
Nevermind, this was a dumb question :(
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles
Sent: Friday, March 09, 2007
Hi,
I'm seeing the following message in the full log:
WARNING[478] asterisk.c: poll returned 0: Bad file descriptor
it's repeated a number of times then I'm disconnected from the running
asterisk instance.
What's the best way to correctly report this?
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
(Technically: It is not part of the ael syntax. #include and #exec are
Off topic:
I usually joke with students about response codes to a SIP bye request:
What happens if you send a BYE and the other side responds 603
declined ?
- I don't want to hangup, I want to continue talking
Mother-in-laws would love that...
/O
Hi,
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and then
the file will be played from start again.
I would like that the play of file is only stopped if the user has pressed the
key 3.
What for an
Hi,
With canreinvite=yes, all the media/rtp traffic for the call typically flows
directly between the two peers. So how is the code in bridge_native_loop
called and when? Is it called and used for any further sip signalling and
not rtp?
Thanks for your prompt reply.
Regards,
Santosh.
Hi,
I
Yifan Zhang wrote:
Hi, all,
I am using Asterisk 1.2.15 with an OpenLine4 card (vpb-driver 4.0). And
Asterisk segfaults. Here is the
output of loading chan_vpb. Very detailed because I turned on vpb
verbose. any lead to solution will be
appreciated. Thanks
This has nothing to do with
William F. Acker WB2FLW +1-303-722-7209 wrote:
Hi all,
I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32
machine, and can't find any problems. So, I decided to upgrade my home
pbx. All went well until I tried using my S101 to talk to the IVR.
Some times, the first
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
Correct.
Does anybody have (or know of) a command line application that would:
) Eliminate pops and other random loud noises.
) Trim leading and trailing silence.
) Trim pauses exceeding x milliseconds to y milliseconds.
) Normalize what's left.
I know about normalize and have figured out how to trim
Hi:
I want to make parking calls easier for my hard-working users. Is there
a way to make the Polycom call park feature work with Asterisk?
In case it just works out of the box, I haven't tried it yet; but the
call park feature isn't enabled on the Polycom phones by default.
-Stephen-
Steve Murphy wrote:
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and then
the file will be played from start again.
I would like that the play of file is only stopped if the user has pressed the
key 3.
What for an command can
Wai Wu wrote:
BTW. We only use Asterisk for a few functions. Everything else is done
on an extenal application controlling Asterisk through AMI.
It's just that a few people have reported stability problems under load
in 1.4.
But if you know exactly what you want and why you're upgrading...
Wai Wu wrote:
I didn't know you are courageous. I upgraded to 1.4 last night.
People are very sensitive about their phones working. *Very* sensitive.
It's hard to be courageous in the face of an angry user.
Let us know how things go.
-Stephen-
___
Am Friday 09 March 2007 22:27 schrieb Time Bandit:
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and
then the file will be played from start again.
I would like that the play of file is only stopped
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
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On Fri, 2007-03-09 at 22:21 +0100, Philipp Kempgen wrote:
Steve Murphy wrote:
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
Davis Sylvester III wrote:
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
If you mean the contents of .conf-file based merged with whatever the
Realtime engine is supplying, I don't think there's a way of seeing both
together.
But you
On Fri, 2007-03-09 at 23:01 +0100, Thomas Winter wrote:
Am Friday 09 March 2007 22:27 schrieb Time Bandit:
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and
then the file will be played from start
Brian Capouch wrote:
Davis Sylvester III wrote:
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
If you mean the contents of .conf-file based merged with whatever the
Realtime engine is supplying, I don't think there's a way of seeing
both
Davis Sylvester III wrote:
Brian Capouch wrote:
Davis Sylvester III wrote:
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
If you mean the contents of .conf-file based merged with whatever the
Realtime engine is supplying, I don't
Thanks for the reply!
Steve Murphy wrote:
At Digium, for instance, we keep all our config files under
SVN, and the config
files are just #exec's for svn checkouts.
Nice.
Just one little mistake I hadn't pointed out earlier; the
extensions.conf would probably really be extensions.ael !
Hi Guys,
Looked at lotsa places on the Web/archives already.
Does anyone have a Makefile for Asterisk 1.4 that
integrates spandsp, app_rxfax, app_txfax?
This patch sure doesn't work with the Asterisk 1.4
Makefile:
Hello List -
I've been slowing growing my extensions.conf file and have been wondering
how everyone manages their systems. I currently have my main
extensions.conf where I reference my sub extensions (for tenants or
customers) files using the include statements and define my global
variables.
Gordon, thanks for such a detailed and full of information email. It helped
me and must have helped hundreds of others on this mailing list.
In my scenario, for this client whom I am working for, their main issue has
always been echo. They have about 50 extensions, with 20 in the office, busy
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get
Hello list,
i'm sure this is not a new issue, i'm having DTMF recognition issues with
TDM404.
I've already tried relaxdtmf=on/off and that did not do any good.
i was wondering if there is any where else in zaptel/zapata to play with and
have it fine tuning.
Or maybe this card is not handeling
Ah!
No, there isn't a chan_zap.so in /usr/lib/asterisk/modules. I installed
over a previous version, however I did delete the contents of
/usr/lib/asterisk/modules before compiling and installing zaptel, libpri
and asterisk.
What is the best way to get chan_zap.so in there? Shouldn't
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