+++ Peter Bowyer [17/04/06 06:57 +0100]:
On 17/04/06, Vikram Rangnekar [EMAIL PROTECTED] wrote:
you can fix issue number 3 by running the install script
sh ./install.sh
or manually running the command
touch /var/log/asterisk/druid
chmod 777 /var/log/asterisk/druid
You'll have
The Problem occurs only on the second card in the System, so i really have no idea.Thanks for your helpNico--
-Ursprüngliche Nachricht-Von: Anthony Rodgers [EMAIL PROTECTED]Gesendet: Friday, 28. Apr 2006 0:24 +0200An: Asterisk Users Mailing List - Non-Commercial Discussion
Opened pseudo zap interface, measuring accuracy...
99.987793% 100.00% 100.00% 100.00% 100.00% 100.00%
100.00%
100.00% 100.00% 99.987793% 100.00% 100.00% 100.00%
100.00% 100.00%
100.00% 100.00% 100.00% 100.00%
Are you sure it should be 4ess switchtype? Have you tried national? Is
it only on 1800/toll free numbers? Pridialplan=unknown, have you tried
anything different for this value?
I can dial into my T3 just fine but I cannot dial out to toll free or
911. Any regular toll call out works,
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have searched google and came up with too many options and packages
that may or may not work for my needs, most articles seem to be for
setting up routers. Maybe someone on the list can give me some
better insight.
I have monitoring turned
On 30/04/06, Steve Totaro [EMAIL PROTECTED] wrote:
My question is, how can I throttle the FTP (Standard with dist)
transfers using out of the box CentOS4.3 (or any easy to use, low
learning curve package)? I thought about FTPing the files at less
frequent intervals but that just makes the issue
Is there a way to monitor the DTMF tones on a channel?
I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.
Is there way to monitor a
On 4/28/06, Matt [EMAIL PROTECTED] wrote:
Well services broke. It's down.. DIDs ring fast busy.Does anyone
know the details of why nufone did not have backup providers? How can
someone lose a contract with a CLEC like that?! Is there more to this
story then we know?
Ok, NOW you can yell
On 4/29/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
I've been playing around with a new system I'm going to install in
another office. In setting up the Polycom's, I accidently used a new
power supply from a new 601 (24VDC) with an 600. The 600 only require
Hi all,
I always get this error message after I hangup a call, what does it mean ?
WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame
cheers,
hn.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
On a side note, I used a 601 adapter on a 501 and the unit failed to
power up. Once I realized the diffence in amps of each power supply I
swaped them and the 501 was fine. It would be nice if the adapters had
some distinction between them.On 4/30/06, Wilson Pickett [EMAIL PROTECTED] wrote:
On
Steve
From what i read here and from what others have suggested i can only
surmise that you tried almost everything besides the simplest thing .
out of the box CentOs installs proftpd (AFAIK ) , this ftp engine has a
simple too called mod_shaper
which allows as you can assume shape traffic
hi,
my asterisk is managing around 500 sip peers, and everytime I do a sip
reload many sip-peers get LAGGED and some get even UNREACHABLE. Any
suggestions ?
cu, florian
--
florian meister
EMAIL: [EMAIL PROTECTED]
TELEPHONE: +43 5572 501 134
FAX: +43 5572 501 97134
ADDRESS:
-- Forwarded message --From: 陈帆 [EMAIL PROTECTED]Date: Apr 30, 2006 10:30 AMSubject: can modify CHAN_SIP.c to generate a new exten= ext,2,dial(tech/peer) ?
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Hi, ALL,
Whether can modify
On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote:
Hi all,
I always get this error message after I hangup a call, what does it mean ?
WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame
This means you hungup while asterisk was trying to play a file to you.
It should be
Original Message
Skype uses iLBC codec, which has great jitter compensation. IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested. I really do not like Skype (prefer FWD), but I must
say, over satellite, etc, they provide
[EMAIL PROTECTED] wrote:
Original Message
Skype uses iLBC codec, which has great jitter compensation. IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested. I really do not like Skype (prefer FWD), but I must
say, over satellite,
What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required in an
implementation that
[EMAIL PROTECTED] wrote:
What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required
Hi,
Red Hat 9.0
Asterisk 1.2.7.1
I'm having a bit of an intermittent problem with my SIP account.
Often (but not always) when I start * or RELOAD my dial plan from the
CLI I get this message:
Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822
add_realm_authentication: Format for
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the
time. Of course his voice quality is like a morse code with dashes or
dots of connection time.
The next minute he calls me via Skype and
Have you tried switchtype=national ?
Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on asterisk. Inbound
calls are working just fine but I am not able to make outbound calls.
Does anyone know what I need to change to make outbound calls work?
Right now the PRI is
I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : Apr
Hi list!
I managed to come reasonably far (farther than I thought I would) but have
two problems.
I still need to pass calls to the Legacy PBX for Fax (I need it as a
channel bank).
I have calls coming in into asterisk, that works fine. Based on the DID I
can route calls to the Legacy PBX
I started with national but then changed it once we looked and the
other pbx was set for 4ess. I'll put it back and look at the debug info again.
At 09:07 AM 4/30/2006, you wrote:
Have you tried switchtype=national ?
Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on
Hello everybody,
does anybody know how to handle the following problem?
I update some gsm audio files every 10 minutes, by rewriting directly on them.
I've noticed that if the file is being played by asterisk exactly in
the moment when I rewrite onto it, who is calling hears a small
jump and
I've installed [EMAIL PROTECTED] and gotten inbound calls going to an extension, extension to extension calling works but I'm still missing a few pieces. The most annoying one is that apparently asterisk is stripping the area code from the number I'm dialing but I can't figure out how to stop it.
I don't know if this helps, from the log.Jim.Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Executing Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new stack
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Deferring dialing... Apr 30 12:58:55
On Sat, 2006-04-29 at 22:49 -0400, Jason A. Kates wrote:
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in
Are you dialing 9 first? It is showing that the digits you
dialed are:
9-770-719-0239
Using your dialplan you should be dialing
1-770-719-0239
Kerry
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL
[EMAIL PROTECTED] (Patrick) writes:
Looks like an SELinux issue. Try booting with selinux=0 or disable
SELinux in /etc/sysconfig/selinux, reboot and see if it works then.
If you to double check it is a SELinux issue, no need to reboot:
'setenforce permissive' will (temporarily) do the trick
No, I'm just dialing 7707190239. When I tried it with a 1, it gave me the same result, a nice lady telling me when making a local call you must first dial the areacode or words to that effect. >From the log, after using the 1:
Apr 30 13:29:04 DEBUG[4242] pbx.c: Function result is ' 7707190069'Apr
Do you have 9 as a prefix in the trunk? It is actually
ADDING a 9 to the phone number before it dials.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim
LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk
Users Mailing List - Non-Commercial
Hi all,
I am running Debian Sarge testing with Kernel 2.6.16.9.
I installed Asterisk 1.2.7.1.
I want to run an AVM Fritz!PCI v2.0 ISDN (rev 02) and an AVM A1 ISDN [Fritz]
card on separate computers.
How can I include them in the new Asterisk ?
Probably mISDN ?
How are they configured at
Hi all,
I am running Debian Sarge testing with Kernel 2.6.16.9.
I installed Asterisk 1.2.7.1.
I want to run an AVM Fritz!PCI v2.0 ISDN (rev 02) and an AVM A1 ISDN [Fritz]
card on separate computers.
How can I include them in the new Asterisk ?
Probably mISDN ?
How are they configured at
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
2) Asterisk times it's outgoing audio based on the incoming
The latency is very high, in that, it picks up after 8 rings. I don't know
what I can tune to reduce to 2 or 3 rings. If it's of any help , I am
posting a section of the log :
Do you get CallerID on that line ?
If, in zapata.conf, you have it set to get the CallerID
(usecallerid=yes) and the
Has anyone managed to add the bristuff patch to
1.2.7.1 successfully?
My attempts has ended up bad, so if anyone has a
working patchfile for 1.2.7.1 I would be grateful to receive it.
Thanks,
Vidar
___
--Bandwidth and Colocation provided by
Calling from a local extension on my local
network, I get good voice quality from asterisk, and
asterisk reliably recognizes my dtmf input.
I set up a sipphone trunk (free) and called in to it
via a separate sipphone account on another computer,
and got slightly lower, but still good, audio
On Sun, 30 Apr 2006, Boris Bakchiev wrote:
I must say, spending just a little extra to get good hardware pays off
in the long run.
If you have any questions, email.
Wow, impressive results must say. Thanks for the specs and test results.
I had hoped that with the Dell 2850 I would
That was it! Thanks.Jim.On 4/30/06, Kerry Garrison
[EMAIL PROTECTED] wrote:
Do you have 9 as a prefix in the trunk? It is actually
ADDING a 9 to the phone number before it dials.
-Kerry
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Jim
LynchSent: Sunday,
Has anyone attempted to use FreePBX for a business in production mode?
Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions. Inability
(or lack of flexibility) is handling multiple incoming pstn lines,
dialplan limitations,
Rich Adamson wrote:
Maybe its just me, but it appears its no where near usable even with the
latest beta1 code.
Its just you. I have FreePBX running on 6 production boxes across the
country. I do very little additional scripting. 5 of the servers have a
Eicon Diva Server V-4BRI card. The
Its just you.
There is much more flexibility on handling incoming pstn lines than there
was in the last version of AMP
If you like manually creating config files with custom settings for each
user, then a GUI is not for you. I have several clients using freePBX
because it is easier to maintain
Rich Adamson wrote:
Has anyone attempted to use FreePBX for a business in production mode?
Yes it works great in business applications.
Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions. Inability
(or lack of
I have uploaded a patch for some manager events that allow to know
when DTMF has been received or sent. Please take a look at this:
http://bugs.digium.com/view.php?id=6082
and if you can, test it and report feedback. Im having problems to
call the attention of bug marshalls for comitting this
Avi Miller wrote:
Rich Adamson wrote:
Maybe its just me, but it appears its no where near usable even with
the latest beta1 code.
Its just you. I have FreePBX running on 6 production boxes across the
country. I do very little additional scripting. 5 of the servers have a
Eicon Diva Server
Rich Adamson wrote:
address zap interfaces, but implies all four lines have to drop into the
same context. Not usable given the above.
The new beta (2.1) allows you to route inbound based on Zap channel --
you could set each channel to route to a specific destination, and
FreePBX will create
Matt wrote:
Is there more to this story then we know?
No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Avi Miller wrote:
Rich Adamson wrote:
address zap interfaces, but implies all four lines have to drop into
the same context. Not usable given the above.
The new beta (2.1) allows you to route inbound based on Zap channel --
you could set each channel to route to a specific destination, and
Rich Adamson wrote:
Well... all those things were installed with FreePBX, they just didn't
grow there. ;)
Honestly, those utilities never been part of FreePBX (nor are they
installed by FreePBX). They are only ever installed as part of
[EMAIL PROTECTED] However, one of the FreePBX developers
You do not say how you have the two connected/
Are you connecting the * to stations via fxo or to lines via fxs on
the legacy?
On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:
Hi list!
I managed to come reasonably far (farther than I thought I would)
but have two problems.
I still
What does you dial command look like?
On Apr 30, 2006, at 12:05 PM, Jim Lynch wrote:
I've installed [EMAIL PROTECTED] and gotten inbound calls going to an
extension, extension to extension calling works but I'm still
missing a few pieces. The most annoying one is that apparently
Ariel Batista wrote:
Rich Adamson wrote:
Has anyone attempted to use FreePBX for a business in production mode?
Yes it works great in business applications.
Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions.
Hermann Wecke wrote:
Matt wrote:
Is there more to this story then we know?
No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.
So why were their services cut? Seems like an obvious omission.
Well... all those things were installed with FreePBX, they just didn't
grow there. ;)
Honestly, those utilities never been part of FreePBX (nor are they
installed by FreePBX). They are only ever installed as part of
[EMAIL PROTECTED]
Actually, they were installed by FreePBX and I still
Hi,
Red Hat 9.0
Asterisk 1.2.7.1
Whenever I start Asterisk, I am unable to call out on my SIP channel:
-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]||t|) in new stack
Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such
host: 6477235412
Apr 30 11:02:01 NOTICE[12814]:
Time Bandit wrote:
Up until this beta1, I could not find a way to support the TDM400 analog
pstn card for incoming calls. For example, pstn line #1 receives normal
business calls, pstn line #2 receives special calls that need to be
routed differently then the context for #1, pstn lines #3 and #4
Rich Adamson wrote:
Actually, they were installed by FreePBX and I still have the iso disk
to prove it
The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an
ISO. FreePBX is simply one of the many software applications that have
been combined to form the [EMAIL PROTECTED]
Rich Adamson wrote:
So, how do you know which conf files one can hand edit versus those that
might be overwritten?
You may only change the *_custom.conf files. :)
--
National Manager - Special Projects
Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore Street T: +61 (0) 3
Also, what is the legacy PBX? On the Merlin Legend, for instance, there are special Class of Services that can be setup to go straight to the auto attendant. I'm not sure if that's what you need or not. The other question is, why can't you transfer the call straight to the extension the fax is on?
Avi Miller wrote:
Rich Adamson wrote:
Actually, they were installed by FreePBX and I still have the iso disk
to prove it
The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an
ISO. FreePBX is simply one of the many software applications that have
been combined to form
Rich Adamson wrote:
zap interface, but apparently undid what existed to edit conf files,
crm, etc. That made things look like a step backwards.
Yeah, a lot of people get confused about that. I was just trying to
clear things up. :)
--
National Manager - Special Projects
Sydney / Melbourne
I am not understanding how queues are supposed to work. I am using
[EMAIL PROTECTED] and configured a queue in AMP. I have also set my static
extensions in the queue. If I set up the system to put people in the queue
on incoming it just hangs up on them. If I try to log in as an agent it says
I am
This is not the right place for help with AAH. Use the AAH forum at sf.net.
If it is just hanging up on users, it is not configured properly.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Patrick Siglin
Sent: Sunday, April 30, 2006 8:25
Duh.. sorry for this dumb mistake, basicaly the connection is:
PRI -TE210 Port 1- * -TE210 Port 2- Legeacy PBX
Basically I need * to send whatever the telco used to send to the pri
Thanks!
On Sun, 30 Apr 2006, Jerry Jones wrote:
You do not say how you have the two connected/
Are you
Thanks! The PBX is a Alcatel Novo Supreme. All calls go straight into the
auto attendant no matter whoch extension I dial on the Zap group the PBX
is connected to. I tried dialling in by hand using several combinations
but I always get the auto attendant.
How do you transfer the call
Hmm... In my case, it could be just dumb luck. I found some instructions on setting up DID on my pbx, and started that. Part way through, I wasn't sure what the rest of the instructions were talking about and felt I was getting in too deep. So, I decided to see what would happen if I just tried
Yes indeed I suspect that * is not passing any DID information on the
call. This could be because my Dial command is wrong or I may need to use
different signalling settings. (Is there any other setting with pri_net?)
When doing pri debug I noticed a line that * was thinking that the other
Thanks alot for the help.
I have not worked on fedra core .Which version should I use
Also can you tell me that if I am using Red hat Enterprise, which asterisk version will be the best suited ? and will i be able to use the same .conf files which i used earlier with aserisk 1.0.3.
I only need to
Hi,
I am about to pull my hair out after trying to get our PRI up and working.
We are switching from a Cisco gateway to an Asterisk box which provides
the 23 phone lines for our office. So, because the Cisco gateway is
working I can assume I have all the settings right (b8zs, esf, dms100,
etc)
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