Bill,
I've had that problem, too. It was caused by too frequent of a
registration and something goofy in the Polycom software. 2.1.2 (the
latest) does not have this problem and I would definitely suggest moving
to it.
It is doubtful that you need that high of a registration period, anyway.
Jay R. Ashworth wrote:
On Thu, Aug 09, 2007 at 11:39:38AM -0400, Gleim, Jason wrote:
I have an Asterisk box connected to a Nortel Option 11C via a T1. In the
Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI
card. The Nortel is also hooked to the PSTN via a T1 on a
[EMAIL PROTECTED] wrote:
Hi,
Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls
just rings and rings. We never get the The number you are trying to reach
If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get
the message on the 1st ring.
Michael Welter wrote:
When I record to a .wav file, I get gsm encoding. Is there a way to
record using u-law encoding?
The extension for ulaw is .ul
Kevin
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To
Keep reading. The person that actually does the calling needs to be
registered. You can't provide the list to others either.
Kevin
Don Fanning wrote:
Oddly enough, there's really nothing stopping one from doing so in the
material I just scan through at:
Response inline.
Steve Totaro wrote:
I receive calls over a T1 with callerid and then *ani*dnis*. I am able
to strip out the ani and the dnis in the dialplan but when I try to set
the caller ID to be the ani, it looks ok but then if I do a NoOp
callerid on the next line, I get unknown.
Hi,
Would anyone be kind enough to send me the 2.0.2 SIP firmware? I asked
VoipSupply for it on Wednesday, nagged them again on Thursday and they
did not even send the request yet. I was supposed to have it 'Friday
morning' at the latest. I'm doing equipment upgrades this weekend so
this
voiplist wrote:
We are aware of the MPG123 tweaks that were always needed with Fedora
in the past. We have MOH working on all other systems.
We just installed a new system with a clean install of 1.2.12.1. It
seems that there is info on the Wiki which states that there is a new
way to do MOH
Tomislav Parčina wrote:
I have queue with member defined as:
member = Agent/SIP/148,1
member = Agent/SIP/143,2
And when I do show queues this is what I see.
pbx*CLI show queues
prodaja has 0 calls (max 5) in 'rrmemory' strategy (0s holdtime), W:10, C:0
, A:1, SL:0.0% within 60s
Martin Joseph wrote:
You mean like setting reinvite and canreinvite to no in your
extensions.conf? This forces asterisk to stay in the media path...
It is only canreinvite, not reinvite! Reinvite is not valid.
Kevin
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Rafael Visser wrote:
Hi gurus...
I have connected an asterisk with a te110p/pri to a GSM ericsson
switch, all apears to be write. But when i try to make an outbond call
from asterisk to the te110p group, the folowing error is logged:
-- Executing Dial(SIP/201-5923,
Simone wrote:
I want to thank you for the suggestions. The office is in the UK, so
probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for
the line so that bandwidth should not be a problem, the internal LAN
will be Gbit as said so the QoS as suggested will be only on the
dovb wrote:
That fix would be great!!!
To press # and be able to get the call back and terminate the transfer...
I had to implement an horrible workaround to emulate this functionality
Well, as it stands now, to hangup while you are doing a transfer, you
using the hangup feature code (in
Barry Flanagan wrote:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
Hall, Eric M. wrote:
[chan_zap.so] = (Zapata Telephony)
Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 setup_zap: Unknown
signalling method 'pri_cpe'
Follow the correct order in installing Asterisk as shown on the download
page at http://www.asterisk.org
zaptel, libpri, asterisk
Kevin
Steven Ringwald wrote:
I am trying to use ImportVar to get some information out of a SIP/ZAP
channel. I cannot seem to find an example of the syntax, or what
variables I can access.
Basically, I would like to output which person is being called. i.e:
SIP/25 calls SIP/21. 25 executes a macro,
nik600 wrote:
after some testing with [EMAIL PROTECTED], i've decided to install my asterisk
server on a slackware (because it's my favourite distro and it is
still suggested here
http://www.voip-info.org/wiki-Asterisk+Linux+Slackware)
, so, i've installed the last 10.2 release, and i've
Anthony Azzopardi wrote:
How come I don't have the MeetMe application registered?
You need a timing source. See:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
Kevin
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Sam Lee wrote:
After installing the timing source , what do I have to do to get meetme
application registered? Do I have to recompile asterisk again ? I don't
see the compiled meetme.so module in the directory.
Yes, you need to compile zaptel, and then recompile/install asterisk.
Kevin
BJ Weschke wrote:
On 1/28/06, Kevin Bockman [EMAIL PROTECTED] wrote:
Joe wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
When an agent receives a call, they will be marked busy anyways as long
as you
Joe wrote:
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
When an agent receives a call, they will be marked busy anyways as long
as you are using agent members for the queue. (member = Agent/1000)
Kevin
Innocent Evil wrote:
To detect an answering machine I have found following two commands,
BackgroundDetect (comes with asterisk)
MachineDetect (asterisk add-ons)
Check out http://bugs.digium.com/view.php?id=5959
app_AMD blows everything else out of the water. I haven't run it in
production
Joseph Rothstein wrote:
I am getting the following error on my PRI-connected Asterisk box,
and am
just wondering if anyone else has seen this, and if so how they solved it.
Jan 18 10:10:23 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event:
HDLC Bad FCS (8) on Primary D-channel of
Dov Bigio wrote:
Hi,
For the dial application, parameter g is described as
a.. g: When the called party hangs up, exit to execute more commands in the current context.
I want the following priority (or at least a priority I can jump to) to be
executed anyway, it doesn't matter which
Carlos Alperin wrote:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
Is anything wrong about this?
I never get 100.00 % most of the times.
No problem there.
David Sampson wrote:
One other question - how do I get outgoing calls to select last
available channel instead of first?
There is an explanation of this in the sample extensions.conf.sample in
/usr/src/asterisk/configs using r/R/g/G
Kevin
___
Eric Lyons wrote:
Feeling once again like an idiot, I come to the list for help...
While my carrier (MCI vi PRI, fwiw) was trying to diagnose a problem,
they asked me to put my interface in loopback -- and I couldn't figure
out what they meant or how to do it. I've plugged a loopback
Chris Mason (Lists) wrote:
When I configured this server, I did not do the make mpg123 option.
Months later, I read about it and did it, as the client was asking about
MOH. About a week later the server crashed, which it never has before. I
believe mpg123 have a memory leak.
What's the best
Dinesh Nair wrote:
we've got a number of TE410P 1st gen firmware cards. could we send them
in to digium for a firmware upgrade ? the cards were purchased in
october 2004 from atp in melbourne.
Yeah, you can. Contact RMA.
Kevin
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Ariel Batista wrote:
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is
to set this dialing rule up.
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion
Cool, I didn't think of that.
Hi all,
I have a couple of LD PRI through Broadwing. I'm trying to verify that
I get the correct cause codes during the hangup. Specifically, I want
to know when a number is disconnected. All of the numbers I have tried
give cause 16. I have gotten a number to give cause 31.
Does
Erik Ginorio wrote:
Now I want to get a DID where someone can call it, have the call
sent to
my Asterisk system via IAX2, and from there have my Asterisk system
route it to the softphone extension that I've got set up. What's the
best way to do this? IAX trunking or just setting up the
Waldo Rubinstein wrote:
One T1 is with one carrier, who provides timing signal.
The other 3 T1s are from a different carrier, all sharing the same
timing signal.
Based on this, I have in /etc/zaptel.conf something like:
span=1,1,0,esf,b8zs
em=1-24
span=2,1,0,esf,b8zs
em=25-48
Dave Morrow wrote:
I just upgraded to the latest CVS HEAD and found that the install reported
app_muxmon.so as being incompatible for this version of Asterisk. Had to
remove it from /var/lib/asterisk/modules in order to get asterisk started.
Yes, the name of the module was changed. The old
[EMAIL PROTECTED] wrote:
I'm thinking about to set up a test environment for a predictive dialler
with two asterisk machines. Each Asterisk should use a Digium TE110P card.
One machine should work as predictive dialler; the other box should simulate
the PSTN.
- Is it in general possible to
Yes, there is -T but it doesn't timestamp everything. All that the OP
posted would not be timestamped.
Kevin
Jorge Merlino wrote:
There is the -T option when running the CLI but I think it only works in 1.2
-- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in new
stack
-- Called
Dave Grey wrote:
I found this pretty interesting -- I didn't know that when you blocked
CID with *67 or per-line blocking that it went anywhere at all.
Apparently, IPKall (who I am using for DID) is doing this
unblocking. I tested a couple of different numbers (A cellphone on a
network
Kevin P. Fleming wrote:
Kevin Bockman wrote:
I agree on both points. I'm not sure if anyone from Digium actually
reads the -users lists though.
Sorry, I shouldn't have said that. I didn't mean it that way. It all
blurs for me too. I didn't think I ever saw you post to -users, but I
guess
Adam Rybak wrote:
s,1,DaeadAGI,test.php,parameter1
How get value of parameter1 in php script?
This is actually a PHP question. You can find it in the PHP manual
online at http://www.php.net
$_SERVER['argv'][1]
Kevin
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KRTorio wrote:
Where in queues.conf? Could you please point out where? Thanks
Check /usr/src/asterisk/configs/queues.conf.sample if you have updated.
Now to state the obvious:
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application,
Waldo Rubinstein wrote:
The only thing I wished was that the Digium cards worked in 3.3V and 5V
motherboards without having to specify which one you are going to
deploy it on. I got somewhat screwed on the TE410P because of that
reason :(
The warranty issue is a big difference. Why
KRTorio wrote:
Is there an easy way to modify the filename of an incoming call's
recording, or are we stuck to agent--unix timestamp format given
to us by Asterisk?
Your answer was in queues.conf that's why you only got 1 reply.
Kevin
___
Patrick wrote:
is there anyway to make the dial command return and execute
the next line in the dial plan after the channel hangs up?
Try with h (for hangup):
exten = 1234,1,Dial...
exten = 1234,h,...
He actually meant the 'h' exten and not priority:
exten = h,1,blah
but that would
Cory Andrews wrote:
Yeah I should have picked up on that, single PCI Riser in this one, so 1
card. I don't know of any 1U solution out there that would give you 3
PCI slots to work with, I think you'll have to go to a 2U at least to
achieve this.
I saw the Dell PowerEdge 1850 has 2 PCI-X on
[EMAIL PROTECTED] wrote:
just as an (bad) example:
we are using an x206 and couldn't get the zttest above 99.975
equal what we were doing
single irq, w/o acpi, w/o apic, different kernels, w/o
hyperthreading, different slots, a.s.o.
for an Digium wildcard TE110P
so if
Tim McKee wrote:
I'm running a large number (125) remote sip phones for FEMA in the Gulf area
over satellite. I've run into a major problem and need some assistance.
When dialing the FEMA voice response system, it appears that it never
actually answers the phone. I never get audio when
Matt wrote:
hmm, I'm not running in safe mode.
looked at /tmp, no sign of core. Linux is strange, there should be a core
file somewhere... i searched the whole / by *core*.
He means by running safe_asterisk. You need to run asterisk with -g at
least to get it to produce a core file. If you
Matthew T. O'Connor wrote:
Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk
1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line.
Nearly every hour, almost on the hour I get this:
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1
successfully
Mark Phillips wrote:
OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back
to V1.09. Other than downloading the code, how do I do it? I thought
someone once said that I have to delete all my modules or something?
rm -rf /usr/include/asterisk/*
rm -rf
Matt Love wrote:
Is there a way to reduce the ring time to answer on the Asterisk @ Home
platform.
Currently it sometimes takes 3 UK Rings before asterisk picks up the call.
In AMP I have Setup-General Settings- Extension of Fax machine DISABLED
Below is my ZAPATA.conf file if this is any help.
amer karim wrote:
I would like to know what's new in motherboard compatible with a
Digium card like: Wildcard TE410P or Wildcard TE110P.
You'll need to specify your platform or say it doesn't matter.
I'm using a 2nd gen TE410P on a SuperMicro X6DHE-XG2 (Dual
Xeon/Video/Dual Gig-E) and on a
Pikoro wrote:
Anyone run into this? This is from the latest 1.2.0 beta1 tarball.
Got it all compiled, but this undefined symbol is stopping asterisk from
loading.
When you change major versions, before you install you should:
rm -rf /usr/lib/asterisk/modules/*
I also rm -rf
Ronald Voermans wrote:
If guess I figured it out already.
I made some changes in chan_sip.c (when ringing was received, it didn't
check for SDP), and recompiled.
I don't know what all of this means, but I'm sure it could be of value
to others. Can you submit your patch to bugs.digium.com?
Arik Funke wrote:
Hi,
following I would like to implement:
1. I receive a call.
2. I hang up the call.
3. I execute a macro
I thought about using call files first... but they don't support macros,
or?
Then I figured I could use php agi after I receive the call, hang up the
call with php
Iqbal wrote:
is there a way of changing the retry time, i.e increase time between
retries, or have more of them
If you mean a call file, check
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Kevin
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Chuck Bunn wrote:
Does anyone know if the Digium Wildcard will work on a PCI Express or
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack
server for use with Asterisk.
They will work in PCI-X of course but not PCI Express. They are
totally different.
You will need the
Tracy Peek wrote:
I have 5 broadvoice accounts and have set up some database gets/puts to
determine of the account is in use when someone needs to make an
outbound call. I have tried adding group=6 in sip.conf and then
dial(sip/G6) in extensions.conf to no avail. Should the group
declaration
Brian McEntire wrote:
My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a
very basic PSTN line coming in from the phone company, I tried to get
the most no-frills line possible (didn't pay for caller ID, voice mail,
etc.). I know I can set up voicemail on * on this line. Can
Steven wrote:
How would I make the dialplan to use a different trunk if the Teliax one is
busy?
This is something I'm testing right now. This is what I use to keep
track of how many channels are in use on each T1 and failover to
something else if there is a problem(?). I'll be using this
Crystal Stream, Incorporated wrote:
I am getting this on the console once people call in
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing
Crystal Stream, Incorporated wrote:
Here is the full transaction
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro(Local/[EMAIL PROTECTED],2,
sipline|3044) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
Hi all,
I hate to ask such a simple question, but it has stumped me over the
past couple of days.
I have 2 asterisk servers connected to the house lan and also via a
crossover ethernet cable. The original purpose of the crossover was to
create a private lan for TDMoE.
I have a TE410P in
Matthew Simpson wrote:
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is hub for ATA
All good central locations with cheap non stop flights.
Atlanta is central for who? With all of the tornados, hurricanes, etc.
I would definately vote no for there. Dallas and
John covici wrote:
OK, I wonder if I have something wrong -- I have the *1 in my
features.conf for the one touch record -- now I called a number, and
when the call was answered flashed the hook and pressed *1 and went
back tothe call, but nothing happpened. I am using CVS from 8/26 --
is this
Lakmal wrote:
/ I have been searching around for days on how to record calls between SIP/
/ Phones. Could someone tell me whether it is possible? The Record command/
/ doesn't seem to work during a call./
If you are using a fairly recent version of HEAD, you can check out Dial
options w
ones.
Thanks,
Kevin Bockman
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Brian C. Fertig wrote:
Try this after your done rotating your log:
asterisk -rx reload
Or just a logger reload... I thought that was already said though.
Kevin
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Jason Walker wrote:
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another
one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver
1.0.7.
That's probably your problem there. I know most newer versions of DIAX
will do this. There is one of the later
Queue + URL and Dial + URL have been in asterisk for a long time (well
before 1.0) so that is not your problem.
Yes, but I'm pretty sure that Queue URL was broken in one of the
previous releases and not fixed until a few months ago.
Kevin
___
Does anyone have details on the “devices” that support the optionalurl
method of the Queue application? I am wondering if there is a softphone
that supports this. The only thing that seems to happen is the queue_log
is updated with whatever is placed in the “optionalurl” location of the
I am trying to update Asterisk from cvs as I think it might solve a
secondary problem that I am experiencing (see below). In the
/usr/src/asterisk directory I typed “make upgrade”. However I get an error:
Makefile:16: *** missing separator. Stop.
Are you on FreeBSD (or not Linux)? You
?
Hope it goes well for you!
Thanks,
Kevin Bockman
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Ryan Stark wrote:
I went to run my queue_log parser so that I could send out a monthly
report to one of my customers, and I noticed that every valid call
complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an
ABANDON:
Here is a complete-caller:
Just a quick question. Does * write directly into PGSQL database like
MySQL?
Theres app_sql_postgres. Uncomment APPS+=app_sql_postgres.so in
apps/Makefile and do a make install.
Kevin
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Yes, Meetme needs timing. You can install ztdummy.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
You also need to recompile asterisk after you compile and install
zaptel.
Kevin
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Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
minutes or so in conference there was a 2 - 3 second delay between the
time
You don't have to use queues to use agents. Do a show application dial
and look at what he is showing you.
You can have a macro run upon answer so put your menu there.
Kevin
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would appreciate it.
Thanks,
Kevin Bockman
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I am happy to tell you that I received a Digium's TDM20B card for my
Asterisk box today. but the phone jack is RJ45 instead of RJ11.
Kumara
You can just plug the RJ11 into the RJ45 jack. It will stick in and
match up.
Kevin
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I've been doing some testing lately on Asterisk. I've had some problems
with it using 100% cpu at times. One time, it held the 100% cpu usage
for 12 seconds.
Are you sure it's asterisk using the cpu? Sounds like mpg123 to me.
If it is mpg123, use madplayer instead and that will
I have the same problem with mine. I think this was something caused by
one of the recent firmware upgrades.
This would definately be something caused by the phone. I don't see
anything in the web config that would set something like this. It
seems like it is some type of flash.
Kevin
8000 -c 1 fle.wav file.gsm, I have the same type of problem.
Thanks,
Kevin Bockman
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the same type of problem.
Kevin Bockman
If I store the file as gsm and record it as gsm, open it was gsm 6.10
with Audacity, it is fine. I know I am doing something wrong with the
u-law conversion.
On a similar note, I have seen in the archives where people have an
output of the file type, showing
gsm and wav files to ulaw. That is my real goal.
Thanks,
Kevin Bockman
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I have figured out that my audio problem was just how I was converting
the sound files. I am trying to convert the Asterisk gsm files to
ULAW.
Kevin Bockman
I have the files in Signed 16bit PCM, little endian, 1 channel, start
offset 1 byte, sample rate 8000hZ. Is this correct so
should be included since I am
running 2.6 and defined USE_RTC.
I checked and /usr/include/linux/rtc.h is there and is the same as the
one from 2.6.11.11 sources.
Thanks,
Kevin Bockman
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make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.11.11'
Building modules, stage 2.
MODPOST
*** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko] undefined!
*** Warning: rtc_control [/usr/src/zaptel/ztdummy.ko]
Hi Tony,
You do need RTC support in the Kernel, because it is the hooks in the
rtc.c driver that the new ztdummy requires.
That's what I thought. That was going to be my next step but I hate
messing with the kernel remotely. I just made it as a module like you
did and it worked. Thanks.
I'm
I'm still having my (apparantly) timing problem, but I'll do some more
testing and make a separate thread for that. I'm generating an
outbound call through Asterisk. The inbound audio is good, but the
outbound audio is sometimes staticy. This seems to happen only at the
start of the call.
From: Asterisk
I have a TE410P running on CentOS 4 which (fortunately) is my test
system. This card has been working flawlessly for the past few weeks.
Today I did a cvs update to the latest head files, and the card is now
not working. When I try a zaptel start, a series of messages
Can it be a problem with AgentCallbackLogin?
-- Executing AgentCallbackLogin(Zap/4-1, 1010|[EMAIL PROTECTED])
in new stack
-- Setting global variable 'AGENTBYCALLERID_XX' to '1010'
-- Playing 'agent-loginok' (language 'en')
-- Callback Agent '1010' logged in on
I have complied an asterisk system and got it going from scratch and all
works great except I cannot make an outbound sip-to-PSTN call and do not
fully understand how to configure it.
Steve Gladden
To get an example setup for your provider, try searching google:
provider
For whatever reason, the music on hold is extremely distorted and loud.
It didn't used to be this way and I haven't changed anything, yet it
persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can
anyone help with this, or has anyone seen this? The mp3s play fine on
any
with CVS HEAD.
-josiah
It does this to me too. We should send in a bug report. I think there
have been other people reporting it also.
It would be reeel nice-like [Beverly Hillbillies] to have this
working.
Kevin Bockman
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What I need to be able to do is immediate answer and play a greeting, then
hang-up. Currently the system answer after two rings #8211; how can I
minimize this?
In /etc/asterisk/zapata.conf:
usecallerid=no
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Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such
inner to this frame (corrupt stack?)
Thanks,
Kevin Bockman
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Darn, I forgot to say that I'm running:
Asterisk CVS-HEAD-04/21/05-16:53:20
FreeBSD 5.3
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Could you tell me how you arrived at the line number of 47 and 92? I
use a few AGIs but it seems to be the vm.agi that it has a problem
with. It is a 67 line file though. :-)
Thanks,
Kevin Bockman
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