when you google the message, you'll find an answer.
jg
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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 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 "Delay Dialing
D
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
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From Line 3, it does not recognize the password.
Did you check whether you have the same DTMF settings for Line 3?
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want to do.
jg
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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 to d
allows for traffic
shaping and a couple of other neat feature, that are usually not part of small firewalls.
jg
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played with this option for quite some time.
jg
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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
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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?
Are t
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 Asterisk.
jg
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
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to execute the ivr extensions.
jg
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themselves.
What I don't understand is why the normal mute button on most headsets is not
sufficient.
jg
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the DTMF features for signaling.
jg
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CDRs while executing hangup logic
This would indicate that at least writing is disabled.
jg
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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 can be
handled here.
jg
be helpful.
jg
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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
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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 from a root
account?
jg
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.
The Redhat script works nicely under CentOS.
jg
list. Very funny.
jg
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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
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=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 problem.
jg
://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 existing
equipment.
jg
calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
Did you study this: http://www.asteriskdocs.org/ ?
jg
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 webaccounts
is to use Action URLs (if supported by the phone) and setup a a finite state
machine externally to handle your needs.
jg
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Asterisk to accept facsimiles, which depends on which
technologies you need. But everything is nicely documented.
jg
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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
and the old Pforzheim files are incomplete.
jg
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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 makes a lot of sense (to me and momentarily).
jg
, 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
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jg
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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
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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
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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
to check your configuration. If
you cannot select an item, there are usually hints on what the resource depends.
jg
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/ ?
Having said that, you might still run into some NAT-related problems, if you
use a normal router.
jg
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|
|
| 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 entries and you would be able to look at the pattern.
jg
,amaflags,userfield,lastdata,uniqueid)
VALUES (blahblahblah, ... ,'1429970147.612')
Can you post the output of describe schema;? At least the first error message is probably
related to a not so optimal primary key definition.
jg
and the product quality is now much better compared to a couple of years ago.
jg
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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 ./configure?
jg
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
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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.
jg
this 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
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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
with yum-priorities. So far
I have not seen any difficulties. Building from source is also very easy.
jg
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Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag.
Then try the rest.
jg
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.
Thank you for your efforts.
jg
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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
predictability does not seem to be a
problem.
Does that make any sense?
jg
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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 problem?
jg
:4569 60 Request Sent
Does someone have an idea at what to look in detail?
jg
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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
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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
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?)
.
jg
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Why are you concerned? ODBC reconnects automatically if necessary.
jg
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It takes a small fraction of second to reconnect. You should not experience any
missing info.
jg
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Unless of course the database server is not running at all for some reason.
But that's not exactly an Asterisk problem.
jg
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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
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So, your DB is not on the same machine? WAN or LAN?
jg
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service provider.
jg
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actually exist.
jg
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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.
jg
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-upgrade.
jg
with 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 athletes.
jg
You asked this question before and there was an already answer on September 28.
jg
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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
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
and Action responses via TCP and Events via multicast?
jg
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Why can't you continue within the extension and dispatch whether the call failed or terminated?
Simply make a second call.
jg
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Did you start the Asterisk server?
jg
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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 call queues.
jg
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 machine.
jg
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 'SIP
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 contact me.
jg
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
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Have you tried RetryDial()?
This way the receiver will be off-hook all the time, which might be
inconvenient.
jg
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Just to be sure, what's the output of vmstat 10 10?
jg
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/telephony-cards/digital/quad-span), it might be worth to
check this.
jg
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if you want to look at it.
jg
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asterisk
Either you do not compile the srtp module into the Asterisk package or you disable RTP
encryption on a phone by phone basis.
jg
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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 ).
How about:
[greeting]
exten= s,1,Answer()
same=n,Background(silence/2hello)
same=n,Wait(3)
provided you know why you want to call Background() instead of Playback().
jg
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What does ps aux | grep asterisk say?
jg
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Wouldn't it make more sense to handle this by just monitoring the calls and doing everything
else with normal data processing?
jg
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What do you mean with voice recorders? Voice mail, if nobody answers, or do
want to monitor calls?
jg
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This wasn't a technical question. It's scam to get some fresh email addresses.
jg
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I, and possibly others, got some unwanted mail from this thread. Somebody is abusing the email
addresses...
jg
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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
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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
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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 the source.
jg
), then there might be a
very small chance to make it work with mISDN.
jg
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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
related to dns, etc...
It is likely that your problem is not related to Asterisk.
jg
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a leading byte with a value of 0xB1 (or
\261 in octal notation). This additional byte is part of the announced string length. Now I
wonder, whether this byte is causing the trouble.
Does anybody know what this leading byte is actually doing there?
jg
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