ed to know what the reason was, but
forgot it. I guess when you google the message, you'll find an answer.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
lines).
Generally, you cannot know whether dialing a number will ring the other end, or not. If all
channels are already occupied for a T1 or E1 connection, the last exchange station will already
signal unavailability, i.e. "user busy" may be signaled by the user or the net
ut
option "c")
But i need this behavior with option c, cause on timeout i need a "Call completed
elsewhere".
How can I achieve this?
Sincerely,
Dominique
Wouldn't it be easier to use a local channel and do something like is done in the
w your "appdata" relate to your previous mails. I am
also wondering why you want to "pass" functions and timeouts. Wouldn't it be enough to dispatch
everything, set some channelvars, assemble a dial string, and then let the local channels take
care of the rest?
jg
--
__
:
From Line 3, it does not recognize the password.
Did you check whether you have the same DTMF settings for Line 3?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
you want to do.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
risk whether a particular endpoint has already active calls and Dial() as
required, i.e. one would delete the phones with active calls from a given list. Since there is
no real "busy" condition, this seems to be a cleaner approach.
At first you should be able
3 superseding the current Asterisk package. If necessary, pfSense allows for traffic
shaping and a couple of other neat feature, that are usually not part of small firewalls.
jg
--
_
-- Bandwidth and Colocation Provided by
o set the auth-tag to AES-80, but I
haven't played with this option for quite some time.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webi
${DIALOPTS})
...
Hangup()
not acceptable? If necessary, one can try to find out which devices are technically available to
avoid dialing a non-existent device. If pressing a "1" is acceptable, then why not pressing the
"DND" to not accept the call?
jg
There's
--
__
AGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()
With this configuration I could send message, but I don't know what wrong with it as sometimes
I get the repeat messages many times. do you have any idea?
moves finished sound files to a file server and
converts them to mp3. The software that accesses the audio looks for both formats at both
places. I think it is generally a good idea to handle file issues outside of
this is related to the settings of silence suppression. I haven't seen this for a while,
but you might want to check the "Silence Suppression", or "Voice Activity Detection" (VAD)
settings of your SIP endpoints.
jg
--
__
for certain
patterns, you'll never get the complete number.
Given the code, there is no reason to execute the ivr extensions.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
y ever find out themselves.
What I don't understand is why the normal mute button on most headsets is not
sufficient.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
could use the DTMF features for signaling.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk
eneral_endbeforehexten>
|Boolean|
|1|
|false|
Don't produce CDRs while executing hangup logic
This would indicate that at least writing is disabled.
jg
--
_
-- Bandwidth and Co
.
Having said that you also need to coordinate your efforts with your telco. You need to check
several transmitting and switching facilities, like CLIP, CLIR, COLP, COLR, possibly CNIP. CLIP
and COLP comes with different flavors. I'd say that the details are outside of what c
t still be helpful.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users maili
rthy
www.asteriskwin32.com hosts only a very very old version of Asterisk (1.2.something). What
speaks against setting up a small virtual machine to host a recent version of Asterisk?
jg
--
_
-- Bandwidth and Colocat
oot is the only account on the machine - which leads me to
believe I have some basic error in my Centos 6.5 installation so I'll just
try it again or try Centos 7.
Okey-dokey. What happens when you start asterisk with "asterisk -c&q
into the /etc/init.d
directory and issue "chkconfig --add asterisk" as well as "chkconfig asterisk on" and your
problem should be solved. You can check the current settings with "chkconfig --list asterisk".
nvalid" to the local deny list. Very funny.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.aste
o differences in the
configuration compared to the rest.
I do not expect that Asterisk is the problem, but does someone know under which circumstances
this kind of problem can occur?
jg
--
_
-- Bandwidth and Colocation Provid
ilbc
allow=g729
allow=g723
allow=gsm
I tried with allow=all, too, but it results in no communication on all
numbers...
Could someone help me?
How is the 4th phone configured?
You could also enable SIP debugging to get more information about the pr
TCDVfMz15M
and https://www.youtube.com/watch?v=1VGkmPF1CNo
Have you ever thought of setting up a virtual machine to (e.g. VirtualBox) for testing and
developing? Most phones allow several SIP accounts, so you could test this with your exist
tus
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this
info, maybe do some action.
Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on
Asterisk because it can affect performance.
02.07.2015, 15:31, "jg" :
Is
calls is to use Action URLs (if supported by the phone) and setup a a finite state
machine externally to handle your needs.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
csimiles, which depends on which
technologies you need. But everything is nicely documented.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
If I call a number from the phone of my wife, I get this warning:
[Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for
gsmtolin
I think this is related to silence suppression. Either ignore it, or find the device that does
this and disable silent suppression.
jg
ow from where you can get
them and the old Pforzheim files are incomplete.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
blem...
So, I don't think, I have to expect problem on my NAT (anymore... initially I
had some problems...).
There's nothing special, only if you want to set up your own infrastructure for
finer control.
jg
--
__
this time. If I look at the complexity of my routers' packet filtering, it makes definitely
sense to separate gateway from internal functionality.
One could say that cascaded Back-to-Back-User-Agents look peculiar, but once you start to think
about maintenance, it
" say?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users maili
(baby
monitors) so you don't have to wait for an answer. Interesting exercise, but might disturb peace
in the house.
If your phone supports only a single identity, then you have to adjust caller ids, etc with
Asterisk.
jg
--
__
n Asterisk?
Yes, it is called "core set verbose 42", the other options is "core set debug
42". Enjoy the show!
Once you are more familiar with *, you might want to have a look what you can
do with logger.conf.
jg
--
Thank you all for valuable input,
another question: when do I actually need the echo cancellation
(hardware / on board /on module ) ?
It depends on your environment. If there are still analog devices in addition to VoIP, I'd say
always, but Asterisk has a rudimentary echo canceller already on
make menuselect" to check your configuration. If
you cannot select an item, there are usually hints on what the resource depends.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
/ ?
Having said that, you might still run into some NAT-related problems, if you
use a normal router.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introd
| datetime | NO | | -00-00 00:00:00 |
|
| uniqueid | varchar(45) | NO | | |
|
...
Just in case you get bogus records with offending primary keys due to some other problem, you
would still have valid data base entri
RT
INTO cdr
(dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid)
VALUES (blahblahblah, ... ,'1429970147.612')
Can you post the output of "describe ;
d the product quality is now much better compared to a couple of years ago.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
I'm not able to install asterisk whenever I hit make command I get below error:
make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'. Stop.
make: *** [main] Error 2
Just guessing. Did you call &qu
uot;? Snom phones allow to define a directory, where you can
export and import a simple text file. There might also be a way to automate this using one of
the provisioning methods.
jg
--
_
-- Bandwidth and Colocation Provid
120
jbimpl= fixed
jbresyncthreshold= 1000
PRI or BRI? Which card are you using? Typically the installation script or procedure lets you
configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels.
is is not an Asterisk feature. It's up to the phone to decide what to do with an
"invitation". There are typically multiple configuration options to take care of questions like
yours.
jg
--
_
-- Bandwidth and Co
update and I am not aware of any stability issues at the moment.
How do you supply power? 3 expansion modules + the phone and a cheap POE switch could be
critical. It may not be the power itself, but the correct handling of energy saving states.
jg
orget about the reverse DNS stuff for the moment.
Do simple SIP accounts (without SRTP/SRTP and deny/permit stuff) work?
Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag.
Then try the rest.
jg
--
_
-- Bandwidth and
but I also work with yum-priorities. So far
I have not seen any difficulties. Building from source is also very easy.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
UDP timeout being too short is another thing I've experience with firewalls (admittedly
limited and once removed experience). Actually, this one can be a (mild) problem on Draytek
routers and can be resolved by telnetting into the router and using the portmaptime command.
Also, turn of stat
blem. Port predictability does not seem to be a
problem.
Does that make any sense?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar e
er vendors.
Thank you for your efforts.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk
: 001 Type: IAX Subclass: ACK
Timestamp: 00014ms SCall: 01200 DCall: 1 79.233.155.174:49153
I am not sure what causes port 4569 to be replaced an an arbitrary port, which could be the
reason for my problem. Does someone know whether this is a router related pro
I found a way that works. Essentially, I deleted the register lines and added the hosts with
deny all and specific permit specs. I don't know why it works, but it does.
jg
--
_
-- Bandwidth and Colocation Provided by
On Thursday 05 Feb 2015, jg wrote:
Calling from ServerB to ServerA works, but not vice versa. The only odd
thing that appears to me is the different perceived port on ServerA.
Does someone have an idea at what to look in detail?
Look in /etc/asterisk/iax.conf in the first instance
ServerB 79.233.yyy.yyy:4569 60 Request Sent
Does someone have an idea at what to look in detail?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory web
OMG.. how embarassing.. that was my personal reminder E-Mail for x-mas dinner. Not meant for
this list. Please ignore. Shame on me.. *blushing* LOL.
Am 12.12.2014 um 21:19 schrieb Markus:
Anna Crepes: Traubenzucker
+ Feldsalat spezielles Dressing (bringt selbst mit?)
Weitergelei
oints.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing li
So, your DB is not on the same machine? WAN or LAN?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
It should not happen. I have a couple of Asterisk servers using the ODBC connection. I never
ever had any problem with ODBC or the database. What database are you using?
jg
--
_
-- Bandwidth and Colocation Provided by http
Unless of course the database server is not running at all for some reason.
But that's not exactly an Asterisk problem.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join u
It takes a small fraction of second to reconnect. You should not experience any
missing info.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Why are you concerned? ODBC reconnects automatically if necessary.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
y
headers actually exist.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-use
communication with your service provider.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/h
Don't they have a kernel module that communicates with the card on one and with DAHDI on the
other side? The first steps are probably to check with lspci whether the card is detected and
then make sure the allo module is loaded.
h the DAHDI
and kernel versions and DAHDI works only if the kernel doesn't object, which happens once in a
while. The wanpipe drivers rarely (=never) work with the most recent kernels, so running a
production machine with the latest beta Fedora or Ubuntu, is something for extreme athl
for kernel updates, but my systems are RedHat based. I
doubt that the Debian based systems are much different as far as interface changes are concerned
for a certain release. My guess is that you should be fine if you do not execute "apt-get
dist
Then this may be the wrong forum. Intercom is also a bit vague---there are a couple of different
options. Have a look at: http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
You might just have to set the auto-answer feature of a phone, but this would
be phone specific.
jg
You asked this question before and there was an already answer on September 28.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Of course, it is possible. Depending on what the desired behavior is, it might suffice to enable
the auto-answer feature of an end point. You might also want to read about paging and intercom
for different scenarios.
jg
Dear all,
My client has Asterisk based telephony system. He needs to
and Action responses via TCP and Events via multicast?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
h
Why can't you continue within the extension and dispatch whether the call failed or terminated?
Simply make a second call.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Jo
Did you start the Asterisk server?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
I would let the phone handle the different ring tones, if possible. For my phones a SIPAddHeader
with something like "Alert-Info: <http://127.0.0.1/Ringer3>" does the trick, but the syntax
might be vendor specific. The other problem should be taken care of with c
If you think it is bad then
do not use it;
else
use it;
There is no natural law that requires to publish the sources, even if the software is otherwise
free. You can always write your own modules and publish the sources. I have difficulties seeing
your point.
Without creating a large
If it's a 64-bit CentOS, then you'll have 64-bit binaries by default. Just compare the size of
the binaries with both options. Years ago there could have been occasional problems, if you had
32-bit and 64-bit binaries on your mac
Please, show your dial plan and name your Asterisk version. You might be call the Dial
application with incomplete arguments.
jg
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with
Have you tried RetryDial()?
This way the receiver will be off-hook all the time, which might be
inconvenient.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Wouldn't the various "call completion" methods require support from the telco? It might be
technology dependent and even for the same technology, e.g. ISDN, telcos might not support or
enable it.
jg
--
_
-
This is a typical phone feature, but it should be easy to implement this at the pbx level using
"originate" and call files. Actually, I have a robust wakeup call module for hotels that could
be used for this. If you need a fast solution you could contac
/telephony-cards/digital/quad-span), it might be worth to
check this.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Just to be sure, what's the output of "vmstat 10 10"?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
know if you want to look at it.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
ast
Either you do not compile the srtp module into the Asterisk package or you disable RTP
encryption on a phone by phone basis.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
The call invitation is only signaled in most cases. You need to check the
settings of your phones.
Hi,
I use Asterisk to create the dial tone (indications.conf), which works quite well. However the
generated signal is quite loud at the client side (in comparison to the following speech ).
Is
How about:
[greeting]
exten=> s,1,Answer()
same=>n,Background(silence/2&hello)
same=>n,Wait(3)
provided you know why you want to call Background() instead of Playback().
jg
--
_
-- Bandwidth and Colocatio
What does "ps aux | grep asterisk" say?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.as
Wouldn't it make more sense to handle this by just monitoring the calls and doing everything
else with normal data processing?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
What do you mean with "voice recorders"? Voice mail, if nobody answers, or do
want to monitor calls?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
I, and possibly others, got some unwanted mail from this thread. Somebody is abusing the email
addresses...
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
This wasn't a technical question. It's scam to get some fresh email addresses.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar e
/configure when building the package fails. Just call "make", maybe
"make --jobs=4".
If your build fails because of a missing library, then you may (need) to call
configure again.
jg
--
_
-- Bandwidth a
Can you publish a short stub of your dial plan to see what you are doing? There are the NoCDR,
ForkCDR, and ResetCDR applications that might help.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
psak could be called from extensions.conf, but I'd like to avoid that. Transferring
to a Local channel after entering the dial plan might also work, but that looks clumsy. I am
sorry if I have overlooked a standard method to send a header back to t
look for delays related to dns, etc...
It is likely that your problem is not related to Asterisk.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webi
I would appreciate if someone could help me with the following issue:
http://pastebin.com/bTskMLVw
My res_odbc.conf file look as follows:
http://pastebin.com/bhReQkXQ
Nothing to really worry about. The ODBC driver automatically reconnects to MySQL as the system
already told you.
jg
), then there might be a
very small chance to make it work with mISDN.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register:
Registration from '' failed for '10.0.1.4' - No matching
peer found
What does the cli command "sip show peers" show? Do you have a definition for the sip de
1 - 100 of 237 matches
Mail list logo
