On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee [EMAIL PROTECTED] wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a
On Jun 8, 2006, at 7:00 PM, Doug Crompton wrote:
I think he clearly states at the end of his message that he is using
the
SPA-3000.
snip
My bad. Should learn to read more carefully (and type too) .
My apologies.
Marty
___
--Bandwidth and
Tzafrir Cohen a écrit :
On Thu, Jun 08, 2006 at 02:12:48PM +0200, Noc Phibee wrote:
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a smartphone
or a hardphone.
What phone is it, exactly?
Thomson
Martin Joseph a écrit :
On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee [EMAIL PROTECTED] wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the
On Jun 8, 2006, at 11:52 PM, Noc Phibee wrote:
Martin Joseph a écrit :
On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote:
On 6/9/06, Noc Phibee [EMAIL PROTECTED] wrote:
anyone have a answer at this question ?
Noc Phibee a écrit :
Hi,
Is it possible de tell asterisk to increase the
Hi:I 've a question:I'm using [EMAIL PROTECTED];
I've seen the dialparties.agi , I want to do this;
I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no;I want have more proxy with others softphone;I run asterisk - R and I've seen when a
Title: Duplicate asterisk processes
I'm still getting duplicate process but the results of gdb are different. Can anyone shed any light onto what is causing this?
(gdb) info threads
1 Thread 1091845040 (LWP 31287) 0xe410 in __kernel_vsyscall ()
(gdb) thread apply all bt
Thread 1
On 9 Jun 2006, at 02:04, Leo Ann Boon wrote:
Jens Vagelpohl wrote:
Hi everyone,
I'm trying to buy a small form-factor PC system for use with
Asterisk and Hylafax in conjunction with a Eicon DIVA Server
single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X
compatible).
Olivier wrote:
2006/6/8, Paul Hales [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Another option would be to see if the provider will provide 2 BRI
lines
that are tied together in some way.
Most of the providers in Australia will do similar things with PRI.
PaulH
Do you
I have made the following additional changes:
- enabling MMX extensions in the Asterisk Makefile and remade, installed
- disabled parallel, serial, and mouse ports in the BIOS
- reenabled ACPI as I was getting errors in the log file
The audio problems still exist. Any further advice on how to
Asterisk Version: 1.2.9.1
Zaptel Version: 1.2.6
LibPri Version: 1.2.3
Hey List,
we are running an asterisk server in connection with an octopus
telephone system. I have expired some random drops of zap channels
bridged to SIP Telefones ( snom 190 ). Asterisk Messages shows something
like
Hi everybody, sorry for my english but i'm italian and i don't know it
very well.
I'm trying to do a java-program to traduce and notify asterisk events to
a Tapi program.
I've a problem with call trasfer.
When i transfer a sip user i would like to put his line on hold but i
can't do it. He listen
Hi,
I've been having problems getting Asterisk to work with a fritz card (in
ptp mode). We've managed to get everything configured, however, as soon
as the call is connected we get Unhandled message: prim 281 len -22 from
addr 100, dinfo 0 on this port
I checked, and mISDN_dsp is loaded.
Have you followed the wiki entry on aussievoip? You are better to use capi
because I don't think misdn actually supports fax.
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 4950 5609
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
-Original Message-
From: [EMAIL
Hi all,
I want to do some stuff in astdb by remotely populating it. That means I
want to make a kind of away-handling. A caller can specify a number on
which he will be available until he resets this. I thought about doing
that with the astdb by setting the the $EXTEN as key and the
to-be-dialed
of chan_misdn ?
I found a bug, and I don't know where to report it.
Andrea
Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
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9 jun 2006 kl. 10.18 skrev Nicola Pascelupo:
Hi everybody, sorry for my english but i'm italian and i don't know it
very well.
I'm trying to do a java-program to traduce and notify asterisk
events to
a Tapi program.
I've a problem with call trasfer.
When i transfer a sip user i would like
9 jun 2006 kl. 10.49 skrev [EMAIL PROTECTED]:
of chan_misdn ?
I found a bug, and I don't know where to report it.
chan_misdn is now part of Asterisk. All bugs in Asterisk are reported
in our issue tracker at http://bugs.digium.com
Christian, the maintainer of the misdn channel, will
Hi,
For some time now, I've been fighting with RxFax and Asterisk.
I had it working for some time, however, for some reason it just stopped
working, I guess someone updated Asterisk or something, don't know exactly.
At the moment I keep getting errors while entering the RxFax stage of a
Hello List
Is there a way to have hints sent between multiple servers?
We are currently implementing a cluster solution for our asterisk servers, and
the problem is this.
User A registers on Asterisk 1 and user B registers on Asterisk 2.
User A subscribes to user B's status, through SIP NOTIFY
Thank you.
I am now trying to put the bug in the best way, according to guidelines.
Andrea
Olle E Johansson
[EMAIL PROTECTED]
ok, it was an extension configuration problem, thanks!
On 6/8/06, Christophorus Laube [EMAIL PROTECTED] wrote:
I do not see the problem exactly. Most of the messages tell about the layer 1
and 2 activation:
P[ 5] MGMT: SSTATUS: L1_ACTIVATED
...
P[ 5] MGMT: SSTATUS: L2_ESTABLISH
First thing
Who is the file who listen when a softphone is run from a remote pc?
-- Registered SIP '651' at 192.168.251.10 port 2209 expires 900
Who makes this in the CLI?
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To
Have a look the sources of misdn... http://www.beronet.com and there
should also be a link to their bug system.
of chan_misdn ?
I found a bug, and I don't know where to report it.
Andrea
Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
Visitate il sito
How can I copy all the contenent of the asterisk database to another machine?
I want copy all the active sessions from one [EMAIL PROTECTED] to another one and running on the second(thisI can do using vrrp protocol, it isn't a problem), I want copy onlyall the active sessions and softphone
Noc Phibee ha scritto:
anyone have a answer at this question ?
I'm pretty sure the answer is you can't, it makes sense to adjust the
gain only where the A/D magic occurs, so you have to tweak your ATAs,
you can set levels in asterisk configs only when configuring devices,
like in
Hello
I can save you a lot of time, and tell you
that it wont work.
It does hold some registration information
in the asterisk database, but most of the information is kept internally in
Asterisk.
Just FYI.
Jon
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne
hello
when I execute asterisk I have this
error
Loaded /usr/lib/asterisk/modules/chan_oss.so
= (OSS Console Channel Driver)Jun 9 11:40:44 NOTICE[5284]:
chan_oss.c:1380 load_module: Unable to load config oss.confJun 9
11:40:44 WARNING[5284]: loader.c:414 __load_resource: chan_oss.so:
ok...but if I run a softphone and it is registered in the CLI and I see this:
-- Registered SIP '655' at 192.168.251.10 port 1175 expires 900
this registration where is put?in which file?
Can I copy this registration to another machine?
On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Its a little more tricky than that.
Our solution involves an external manager
application, some clever IAX2 routing and dialplan mysql queries.
We tried the solution with just copying
the registration, but it seems as though the SIP channel has the registry
information in an
Internal
I've read your post on the asterisk mailing list. Agree that the specs
of that box should easily handle one call with decent quality. The only
thing I can think of right now is to start using the IRQ affinity stuff
to move the scsi ethernet modules over to e.g. CPU2 and let the wctdm
driver
Somy only solution is to use only X-lite softphone where I can add more than 1 proxy, and a Cisco switchboard where I can set up a VRRP protocol, so in case of fall, the cisco make the resolutions of all tables and permited me to call from IP phones like CISCO IP phones or wi_fi phone without
Hi,
Can anybody tell me, is their a tsp for asterisk on linux
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There is a solution, but its not straight
forward, and not really documented anywhere.
A possible solution, is to set a SER
server up, before your asterisk, and let that handle the SIP registrations.
Jon
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Shenen
Hello,
Is there a project for SRTP/SIPS in Asterisk ?
Harry
__
Do You Yahoo!?
En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible
contre les messages non sollicités
http://mail.yahoo.fr Yahoo! Mail
Yes
There is AstTapi:
http://www.voip-info.org/wiki/view/AstTapi
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af [EMAIL PROTECTED]
Sendt: 9. juni 2006 12:32
Til: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Emne: [Asterisk-Users] TSP on
Do you mean receiving traffic on 2 BRI lines (2 channels spread on 2
separate ports) connected to 2 differents boxes so that one line or
box
failure wouldn't affect incoming calls ?
If positive, do these providers price this service (2 ports - 2
channels)
at an intermediate level between
Hi all,
I've just acquired a few Motorola OJO units, and am looking at testing them
out with Asterisk.
Having read the Ojo spec it seems they use an 'enhanced' version of h.264,
which I believe is supported to some extent under 1.4 (?). Amongst other
things, it seems that the Ojo allows for
On 6/8/06, Brian Swan [EMAIL PROTECTED] wrote:
[snip]
I've followed the numerous suggestions in the mailing list archives
which is what has enabled me to get this far. After trying all the
echo cancelers, and all the settings on each I settled on:
- KB1 (with AGGRESSIVE_SUPRESSOR)
-
Hi,
Can anybody tell me, is their tsp for asterisk on linux
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Christophorus Laube wrote:
But I want to set the number remotely and client initiated. Is AGI able
to do such things or what can I use else?
Have a look at the Manager API and the DbPut action for this.
http://www.voip-info.org/wiki/view/Asterisk+manager+API
Hi
this is the configuration i will realize:
i configure the pickup feature
pickupexten = *8 ; Configure the pickup extension. Default is *8
then i have a group that contains 5 SIP extensions
for example
SIP/100
SIP/101
SIP/102
SIP/103
SIP/105
i would like that all the calls are forwarded
Brian Swan wrote:
I've spent the last week or so troubleshooting echo problems at my
Wife's business, and I've been able to clear up about 99% of the echo,
but there is still a little residual echo that I can't seem to tweak
out. The users describe it as buzzing or crackling, but what it
I have set the name to before transferring and it doesn't work.
The PRI debug still shows that it is trying to send .
--
--
Steven
http://www.glimasoutheast.org
trixter aka Bret McDanel [EMAIL PROTECTED] wrote in message news:[EMAIL
PROTECTED]
On Fri, 2006-06-09 at 07:47 -0400, Steven wrote:
I have set the name to before transferring and it doesn't work.
The PRI debug still shows that it is trying to send .
Perhaps try:
exten = 123,1,Set(CALLERID(name)=)
Regards,
Patrick
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Stewart Nelson wrote:
I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems
there. Calling in, though, the phone doesn't ring. Caller ID shows
up, I
can pick up the phone, and the call is connected, but no ring. I've
tried
it on two analog phones, same behavior.
I have tried that as well.
I will do a debug of:
hidecallerid=yes
hidecallerid=no with a call from telco with no callerID.
hidecallerid=no with a call with callerID.
--
--
Steven
http://www.glimasoutheast.org
Patrick [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
On Fri,
On 6/9/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
Consider getting a Sangoma A200D
(http://www.sangoma.com/datasheets/p_a200-specs) with the optional
hardware echo canceller module. It just works for echo cancellation;
no tweaks required. It takes a while to figure out how to install
Guys.
I have a couple of agis that when trying to dial a local call, LD, etc. ask
the user for a password and then checks against a DB to see if they can call
or not.
My newi dea here is to allow users to roam between extensions, for example,
user 1 can go to users 2 phone and when ask for the
Who mentioned fax? :-)
Native CAPI isn't an option for us (unfortunately), as our service is a
point-to-point service, which the AVM CAPI drivers don't support.
We've now got another problem -- we're now able to make calls -- once. The
kernel panics as soon as someone terminates a call (and,
Hi there,anyone in the community has manage to configure click to call features? Care to share.I have tried on this manual , seem got some software error like
http://www.voip-info.org/wiki/view/Asterisk+click+to+callSoftware error:
Unable to determine call statusMessage: Originate with 'Exten'
I have been playing around with sipX for a couple of days now, and while
I don't really like it (just feels wierd), I do really like the
management interface for provisioning phones. I was wondering if anyone
had considered ripping this out of sipX or porting it to a simple php
interface or
hi all i have my asterisk work perfectly thanks to you now i need to apply some improvements like PIN :(i need to all people can call in my city with this patternexten = _XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _XXX,2,Voicemail(u${EXTEN})this work perfectly but when the people try to call a
Pietro U wrote:
hi all i have my asterisk work perfectly thanks to you now i need to
apply some improvements like PIN :(
i need to add somenthing in here or in other section in the conf?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Authenticate
Doug
--
Ben Franklin quote:
What kernel are you using Chris?
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 4950 5609
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Jones
Sent: Friday, 9 June 2006 10:36 PM
To:
Callum McGillivray wrote:
Haha - My director wants to do away with the analogue line that we have
for our fax machine at the moment !! I really don't want a whole PBX
sitting around just to handle a single fax machine
Thanks for the info / experience though
Has anyone else got any
My design is the following:
SS7_smgd[asterisk]Zap-backtoback---EuroISDN[Mediatrix box]Sip
customers
In this design I have a problem about early audio
no passed from asterisk to Zap un the incominga cal from Zap.
I am sure SS7_smgd passes "early audio" to Asterisk
because if I
On Thursday 08 June 2006 23:08, Callum McGillivray wrote:
We currently have an ISDN 30 installed on a Dell 2850, using a single
port Digium ISDN card. We would like to add an analogue card, plug our
fax machine in and have Asterisk simply detect the fax and pass it
through to the fax machine
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?
Thanks,
Daniel
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I did the debugs and posted them at http://bugs.digium.com/view.php?id=7321
--
--
Steven
http://www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
I have tried that as well.
I will do a debug of:
hidecallerid=yes
hidecallerid=no with a call from
An update about my issue is that SS7_smgd still not
pass cause code IE from PSTN.
should it be the cause of the not flow of early
media?
Regards
Rosario
- Original Message -
From:
Rosario Pingaro
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent:
good question! I'd like to know too, so keep it public please !:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Friday, June 09, 2006 9:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000 MultiPurpose
On 6/9/06, Doug Lytle [EMAIL PROTECTED] wrote:
Pietro U wrote: hi all i have my asterisk work perfectly thanks to you now i need to apply some improvements like PIN :( i need to add somenthing in here or in other section in the conf?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
On Thursday 08 June 2006 23:08, Callum McGillivray wrote:
We would like to add an analogue card, plug
our fax machine in and have Asterisk simply detect the fax
and pass it
through to the fax machine
This is what we use to play a beep every 30 seconds on our calls (it's
for monitoring purposes)
I know that SetVar is deprecated but i'm too lazy to change it. This
works great for us on 1.2.4
exten = s,n,SetVar(LIMIT_WARNING_FILE=beep_quiet)
exten = s,n,SetVar(LIMIT_PLAYAUDIO_CALLEE=yes)
C'mon guys! Certify a few current
model servers and be done with it.
Problem is, certification is a
moving target and can become invalid with something as simple as a BIOS change
by the manufacturer.Now that the barrier to entry to changing a design
is almost nil,
make sure you have disabled irqbalance - i didn't see anywhere where
you had, so i'm assuming you haven't...
irqbalance will automatically redistribute the IRQ affinity to
whatever CPU has the lowest usage...for the most part, it shouldn't
hurt anything, but i'm with you on making sure the cards
What you are proposing is quiet simple, and is done regularly. We
provision Linksys Sipura ATAs via a perl script with SSL and client
certificate authentication, as well as Polycom phones via XML file
drops. The newest Polycom firmware also states that ssl is supported,
but we have not made the
I
have, using Active Server Pages + Flash. See: http://new.landmarkmasterbuilder.com
and click on Contact Call Us Online. I can post the .asp and .fla somewhere
if someone is interested in it.
-Original Message-From: Sharon Lim
[mailto:[EMAIL PROTECTED]Sent: Friday, June 09,
- channel bank(s) with an attached fax machine
- combination of iaxmodem and hylafax
I am using both with great success. My measured failure rate this week on
inbound faxes using IAXModem + HylaFAX is .003% with just under a thousand
recieved. I found the trick with IAXModem is to offload fax RX
This may not be the most gracefull approach, but this is how I have done it.
exten = s,n,Setvar(Auth=No)exten = s,n,DigitTimeout(3)exten = s,n,ResponseTimeout(5)exten = s,n,Read(MyPin,agent-pass)exten = s,n,Gotoif($[${MyPin} = ${GOODPIN}]?n:n+x)exten = s,n,Background(pin-number-accepted)
exten =
immediate=no
immediate = dialtone
Don Pobanz
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On further investigation, the SMP affinity on ALL of the IRQs is set to
0001. This implies that everything is handled by CPU0, which it
clearly
is not!
I'll do some more research on this but in the meantime if anyone has any
advice on this issue I would appreciate it.
There is a kernel
Here is an example of a dialplan that
looks up a post dial code in a mysql database and updates the accountcode
accordingly.
exten =
_1XX,1,Read(postcode|beep|3||1|10)
exten =
_1XX,2,set(level_auth=0)
exten =
Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file?
- Erick Perez [EMAIL PROTECTED] wrote:
When is supposed to be released the Zimbra+Asterisk version?
There is no 'Zimbra+Asterisk' version of anything. There is a Zimbra connector
(called a Zimlet) under development that allows Zimbra users to place calls via
an Asterisk server, but you'd
I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being sent.
exten =
- Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Is there a way to replicate subscription info between asterisk
servers?
Not at this time, no. That will be probably be worked on during the next
development cycle.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
- Adam Linford [EMAIL PROTECTED] wrote:
Does anyone know whether h.264 will support these units or not? Anyone
with
any experience getting them working?
Asterisk's support for H.264 is only passthrough and file writing/reading, so
it pays no attention to bitrates or framerates.
--
Damon Estep wrote:
What you are proposing is quiet simple, and is done regularly. We
provision Linksys Sipura ATAs via a perl script with SSL and client
certificate authentication, as well as Polycom phones via XML file
drops. The newest Polycom firmware also states that ssl is supported,
but
Pietro U wrote:
On 6/9/06, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Pietro U wrote:
hi all i have my asterisk work perfectly thanks to you now i need to
apply some improvements like PIN :(
i need to add somenthing in here or in other section in the
Actually, that's what I started out with, and outboud calls were the
same as now, inbound calls had a huge amount of echo (until I turned
on Aggressive). In my testing I actually didn't notice any
difference between KB1 actually worked better then MG2 thanks
for the advice, though!
Hello All ,
On Fri, 9 Jun 2006, Gonzalo Servat wrote:
On 6/9/06, Joshua Colp [EMAIL PROTECTED] wrote:
[..snip..]
I'd just like to note that AEL2 was brought over into Asterisk trunk
(what will become 1.4) and the old AEL removed. That's where most
development is taking place on AEL2,
Every once in a while one of my FXO interfaces will not accept
incoming calls. The other interfaces on the same board will continue
to accept calls.
Here is what I see in the logs:
ERROR[4465]: fsk_serie made mylen 0 (-22)
WARNING[4465]: CallerID feed failed: Success
WARNING[4465]: CallerID
Yes you can if you are running 1.0.2.13 or later. I have the template
which I tried posting here as an attachment but it has not arrived yet.
If it does not arrive you can email me directly or contact grandstream
support.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to
Hello,
I've install Rapid for try to solve my problems of external acces (i
can't receive and send calls).
I nedd some more information:
On a Bristuff what are the channels names??
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i
Does it matter if you use upper or lowercase rules - I.E. - x vs. X or
mix them? Not that I would do that as a rule but sometimes you make
mistakes!
Doug
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
*
(Zap/1-1,
recordingcheck|20060609-095557|1149864957.408) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060609-095557|1149864957.408: Inbound
recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(Zap/1-1
I can call/receive fine from a zap fxs connected phone to my voip
provider through asterisk. I can call from a sip softphone extension to
the zap extension just fine. However using a sip phone extension
connected to asterisk and calling out through the voip line there is no
sound. I was able
Hi,
often on this list I read about transcoding as the heaviest activity for an
Asterisk server, together with high IRQ rate (especially with Digium
cards...).
Is there a way to monitor if Asterisk is engaged (by mistake or by design)
in transcoding or any other heavy activity?
Or a checklist to
Can you then inform me on what structures this information is stored in, in the
asterisk code? Then ill try to do a quick dirty version of the replication.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Kevin P. Fleming
Sendt: 9. juni 2006 16:25
Hi Enrico,
This is a perfect question for Digium support. Just call them and
complain that the card is acting like this.
I was struggling with the same problem about a year ago. Managed to get
my card and FXO modules replaced and that helped a lot.
My problem is still happening like once
Hi,
Is there anyway to customize the ring tones on an aastra phone?
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Hello,
We are planning to biuld a 100 lines
PBX based on asterisk.
How do you decide on the system config,
e.i motherboard, cpu , how much ram , etc ?
Is there any thumb rule ?
Thanks
Varun
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Wow, I missunderstood a zimbra article I read somewhere.
A zimletthanks.
On 6/9/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Erick Perez [EMAIL PROTECTED] wrote:
When is supposed to be released the Zimbra+Asterisk version?
There is no 'Zimbra+Asterisk' version of anything. There
Hi,
I am fairly new at working with Asterisk.
I am having a call quality issue that I really need to get
ironed out before we go to rollout the system in a week.
Any help would be greatly appreciated!!! Even if it is just
pointing me in the right direction.
My current setup:
I have
On Fri, Jun 09, 2006 at 04:50:11PM +0200, Olivier Saulnier wrote:
Hello,
I've install Rapid for try to solve my problems of external acces (i
can't receive and send calls).
I nedd some more information:
On a Bristuff what are the channels names??
For outgoing call, i name the channel
-record-enable,s,4)
-- Executing AGI(Zap/1-1,
recordingcheck|20060609-095557|1149864957.408) in new
stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060609-095557|1149864957.408: Inbound
recording not enabled
-- AGI Script recordingcheck completed
I did this and still no dialtone.
Gene Brown
On Fri, 2006-06-09 at 09:23 -0500, Don Pobanz wrote:
immediate=no
immediate = dialtone
Don Pobanz
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To
I was trying to modify some rules in my [EMAIL PROTECTED] config this morning,
and I
was having a LOT of trouble, and not understanding why it was ignoring
some outbound rules, and it wasn't 'till I made all my NXXNXX type
characters all uppercase that it worked properly. I didn't think it
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