, please help. What am I missing?
Sorry for my bad English.
Regards,
Markus
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On Tue, 2009-03-10 at 15:44 -0500, Kevin P. Fleming wrote:
markus wrote:
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
You 'failed' because you installed Asterisk 1.6.0.6, which contains a
very large
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command (which I
made up).
So rather something like:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}ww))
_X.,3,ConnectLegs
Hi,
new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?
I'm using
Hi again,
no one got an idea? :-( Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)
Thanks again!
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command
Here's a script to call a bunch of numbers (list-of-numbers.txt), trigger a
new call every 10 seconds. Adjust for your needs:
-snip-
#!/bin/bash
for i in `cat list-of-numbers.txt`
do
echo /usr/sbin/asterisk -rx originate local/@from-local extension
$i@voipout
/usr/sbin/asterisk -rx originate
Can't help you with the SIP account but for geographical numbers in all
3 countries that you mentioned try http://www.globalnumbers.de -
forwarding to any SIP destination is free.
Am 01.02.2012 13:29, schrieb Christian Gansberger:
Hello List!
I'm searching for SIP-Providers in the following
But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line.
:)
Am 09.02.2012 15:59, schrieb Bryant Zimmerman:
We designed our solution the following way.
We have
Reply to self, missed the line count part. Nevermind then :)
Am 09.02.2012 18:10, schrieb Markus:
But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line
Am 07.03.2012 02:04, schrieb Mike Diehl:
I tried the chat as well with no effect. My German is a bit rusty, or I'd call
them
Most Germans speak English. :)
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Am 14.03.2012 00:34, schrieb James Sharp:
ping + arp isn't going to work if they're on a different VLAN.
I believe this will work:
Too complicated. Just have a look on the switch(es) the phones are
connected to.
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Am 15.03.2012 00:35, schrieb Benny Amorsen:
Markusunive...@truemetal.org writes:
Does such a thing exist?
How does a2billing do it? It should be pretty easy in an AGI. If you can
afford a linear lookup per call, just grep through the array of prefixes
to find the ones matching a particular
all the time on FreeSwitch Freenode channel and it probably
does on OpenSIPs as well. Check there for some good advice.
On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen
benny+use...@amorsen.dk mailto:benny%2buse...@amorsen.dk wrote:
Markus unive...@truemetal.org mailto:unive
Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर):
On Thursday 15 Mar 2012, Markus wrote:
With like 10 different ratesheets from 10 different providers, of
which many change their rates every few days, manually doing it in
Excel is too time consuming...
Is it possible to get samples? I'd
Am 16.03.2012 04:14, schrieb Ast Coder:
I would be more interested in a system where quality routes are tested
with different providers because rate really doesn't matter if a call
can't be placed or if a destination is a fake one. We have seen many
fake destinations with top tier providers but
I hope this is not too off-topic. As a kind-of follow up to rate sheet
normalization I'm still trying to figure out a solution for: throw 10
ratesheets at a program and get the cheapest codes/providers as output.
For this purpose I believe I need a real, detailed, accurate list of all
the
know once the list is ready.
I've already registered opennumberingplan.org :)
Am 29.03.2012 11:12, schrieb Lenz Emilitri:
DO you know if the doc files from the ITU are available somewhere for
download?
l.
2012/3/22 Markus unive...@truemetal.org mailto:unive...@truemetal.org
I hope
Am 28.03.2012 20:17, schrieb C. Savinovich:
The Way to make money is to help folks use the open source items
in the most efficient manner
Nobody wants to pay me $2,000 to install and configure A2billing, which
in my view, is a fairly low price for my time. There are people who do
that for
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
Log the actual DTMF to your console, set in
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
PS: You are only allowing the GSM codec for your
.
PS: The peer doesn't support ulaw.
PPS: Asterisk 10.7.0
Thanks a lot!
Markus
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,
but just one ] closing? Is that a typo?
Also, the doc shows:
GotoIf(condition?[label1]:label2)
Why is label1 in square brackets and label2 isn't?
I'm confused. :)
Thanks so much!
Markus
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? There are two opening square brackets, but just
one ] closing? Is that a typo?
Also, the doc shows:
GotoIf(condition?[label1]:label2)
Why is label1 in square brackets and label2 isn't?
I'm confused. :)
Thanks so much!
Markus
---
You need to run your logic the other way. What you're doing now
on the voice quality but the messages
on the console are quite annoying.
PS: The peer doesn't support ulaw.
PPS: Asterisk 10.7.0
Thanks a lot!
Markus
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Hi Larry,
Am 26.08.2012 03:57, schrieb Larry Moore:
On 26/08/2012 3:45 AM, Markus wrote:
When I receive an incoming call from a SIP peer where I've configured
disallow=all
allow=alaw
(and no other codec)
I can see the following NOTICE on the console:
Dropping incompatible voice frame SIP
puzzled. :)
Regards
Markus
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Background music during a call
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg254252.html
Does anyone have the right solution and is available to create a
dialplan for me for cash? Please get in touch!
Thank you!
Markus
of misunderstanding here. Maybe I'm not
explaining it right...
Thanks!
Markus
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:10380
/var/lib/asterisk/mohstream-chatfire-2.sh:
#!/bin/bash
/usr/bin/mpg123 -q -r 8000 -b 0 --mono -s http://stream.laut.fm/chat-fire
That's it :)
Regards
Markus
Am 27.08.2012 21:29, schrieb Matthew Jordan:
- Original Message -
From: Markus unive...@truemetal.org
To: Asterisk
Dear list,
sometimes my network receives a large DDoS (packet flood, attack) that I
manually block after a few minutes or it just stops by itself. As soon
as the attack stops, the network is fine again, ping is back to normal,
traffic levels fine again, enough bandwidth etc. During the DDoS
and replace
the vm-rec-name file?
Thanks!
Markus
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,Hangup()
Help! :)
Thank you,
Markus
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Am 26.09.2012 17:53, schrieb Mark Robinson:
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine
to send or receive faxes. We are planning to have a dedicated did
, it's free to use.
http://das-asterisk-buch.de/faxserver-mit-iaxmodem-und-hylafax.html
Regards
Markus
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It's an all-SIP scenario with RFC2833 as the DTMF protocol.
Is this a known bug?
Thank you!
Markus
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Thank you!
Markus
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but Asterisk doesn't seem to recognize them (which is
fine as not all providers support all DTMF variants).
My resume is: DTMF is just fine, ConfBridge dtmf_passthrough is not
working at all. Agree? :)
Thank you,
Markus
(123)
(X-Lite tests only with force inband YES, RTP 2833 YES)
If I understand right, all my four DID providers are broken?
Thank you!
Markus
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New
. Is there nothing I can do to remove DTMF tones from my
conferences? :-(
Thank you!
Markus
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Am 29.09.2012 10:49, schrieb resea...@businesstz.com:
[tz-ivr01 ~]# uptime
11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57
Sharing is caring
Is that a Quad Core CPU in your box?
PS: Yes, Asterisk is great. :)
--
Am 29.09.2012 20:17, schrieb Mitch Claborn:
Sam - can you send output from a top when your server is under load?
Just curious.
Preferably with all CPUs showing (hit 1 in top) - thanks! :)
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Am 01.10.2012 22:13, schrieb Eric Smith:
I have the following in sip.conf for asterisk running on localhost.
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
Really on localhost only, no peers other than from 127.0.0.1? Not sure
why deny/permit is not working, but you could do
Am 01.10.2012 20:43, schrieb motty.cruz:
I can't find a clear procedure to lower musicOnHold volume!
Any suggestions?
exten = 1234,n,Set(VOLUME(TX)=-3)
exten = 1234,n,MusicOnHold(default)
in your extensions.conf should do the trick. (play around with the -3 value)
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Am 16.10.2012 18:38, schrieb Sahil Gupta:
The provider appears to be running PortaSIP. Anyone with suggestions?
What does your register = line look like in sip.conf? (without the
password)
What does sip show registry show?
What do you see on the Asterisk console (asterisk -vr) when you
Am 11.11.2012 11:46, schrieb Eric Kuhnke:
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
Miguel already explained what's going on. Have a look at the SIP packets
to
if there are active callers, of course.
Is there any way to restart the MOH system without restarting Asterisk?
Asterisk 10.7.1 and 10.8.0.
Thanks!
Markus
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Am 13.11.2012 16:51, schrieb OCEANET - Cédric BASSAGET:
have you tried module reload res_musiconhold.so ?
Hi Cedric,
thanks for the suggestion. Unfortunately, it does nothing, just like
moh reload.
Any other suggestions?
Regards
Markus
an Asterisk restart.
When I try to unload the module I get:
loader.c:542 ast_unload_resource: Soft unload failed,
'res_musiconhold.so' has use count 2
Is there a way to force the unloading?
Any other suggestions?
Thank you!
Markus
the RTP IP in SDP automatically
right? But I'm just guessing...
Here are some instructions for multiple instances:
http://forums.asterisk.org/viewtopic.php?f=1t=71510
Regards
Markus
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Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call
not a programmer or I
wouldn't have chosen shellscript. :)
You can get it here:
http://sourceforge.net/projects/voipgmap/files/
And if you like it and decide to use the output on the web somewhere,
please let me know the URL so that I can check it out, thanks!
Regards
Markus
in Contact: when
sending its replies?
Thank you!
Markus
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easier than what I came up with, so I'd recommend to Markus
that he try your suggestion first.
bingo, that fixed it! Now everything's working fine, and my config looks
like this:
host=1.1.1.1
type=peer
insecure=port
nat=no
Thanks a lot!
Although I have to say I don't understand what is going
Am 23.05.2013 15:14, schrieb Marie Fischer:
For voice calls, you could try Skype Connect, which is SIP - but needs a
business account, so not free. http://www.skype.com/en/features/skype-connect/
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free.
Am 23.05.2013 16:04, schrieb Richard Kenner:
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)
www.mhspot.com/sts/
(site is down right now)
And that's related to the problem with it: it hasn't been maintained for
quite a while.
True, but it's
Help!
I have providers configured that send me incoming calls via SIP. There's
one that seems to make trouble. As soon as I get a few concurrent calls
through this peer, Asterisk CPU load goes up to 250%, audio becomes
laggy and I get hundreds of these per second in the logs:
[Jun 20
so much! :)
Regards
Markus
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asterisk-users
-with-dynamic-email-recipients/
HTH
Markus
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Am 03.09.2013 02:30, schrieb neo haux:
I want to recieve calls to my Skype account and forward them to a
SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it
only available for asterisk 10
I know that Digium gives no support for this module, but I am sure that
someone
Am 04.09.2013 15:36, schrieb Zyumbilev, Peter:
I searched a lot last few days but I am uanble to find a DID number in
Macedoania.
However no luck. any ideas about a provider ?
didlogic.com had some a couple months ago, but they only lasted for a
few weeks, probably offered by an individual
what exactly is causing that high CPU load?
Thank you!
Markus
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but it helps
with hanging audio streams...
When using several moh file classes, I have never hat audio problems.
Me neither.
Thanks!
Markus
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Am 20.09.2013 16:37, schrieb jg:
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams.
I forgot something. Even if your 7 streams are mp3 streams they cannot
consume the CPU power you are seeing.
I thought so. So, I need to dig deeper... but how?
Some command to show the
triggered like this on outbound calls:
Started music on hold, class 'default', on
SIP/outbound-sip-provider-0002
I do not have any reference to MusicOnHold in my extensions.conf so a
misconfiguration is unlikely.
Is there some SIP magic that can trigger MusicOnHold on my end?
Thanks!
Markus
; echo; echo; echo; sleep 1; done log.txt
Regards
Markus
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Am 08.01.2014 16:07, schrieb Jonas Kellens:
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Look at session-timers in sip.conf. I had to set it to refuse for a
specific
Am 16.02.2014 03:30, schrieb Nick Cameo:
Tried setting `sendrpid = yes` and still same problem. We really don't want to
have to `SipAddHeader` as it is already being formed by our switch.
From http://www.voip-info.org/wiki/view/Asterisk+SIP+trustrpid :
-snip-
P-Asserted-Identity
Asterisk
, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1d46fec6
Content-Length: 0
I'm thinking the answer is no, but is there any option how I can get
the remote SIP provider to answer me on port 5060? Without having them
to change anything in their config.
Thank you!
Markus
Am 20.02.2014 19:48, schrieb Alex Villacís Lasso:
My concern is that asterisk is left listening for SIP through all
interfaces and with no SIP passwords. I want to secure the setup against
directed traffic to the asterisk UDP port (5080), that bypasses the
kamailio process. I tried setting
such an option exists, I just haven't found it yet? :)
Thank you!
Markus
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.
Thank you!
Markus
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feasible, and
switching to another country is not at option either. :)
All is good now!
Thanks!
Markus
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-frontend out-of-the-box, unlike OpenStack. So
you could offer your customers a self-managed, redundant Asterisk cloud
or something like that. :)
In theory, this combination should give you a 100% redundant,
auto-healing, auto-scaling VoIP setup. :)
Regards
Markus
DTMF. These callers use their mobile phones to dial in. I just
reread the Wikipedia article on DTMF but I don't understand how someone
can send an 'A'. Any clue?
Thank you!
Markus
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Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain:
New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.
Hmm. Could this have to do with session-timers (sip.conf)?
I remember when I went from 1.4 to 10.7 I had to manually mess with
Am 26.08.2014 16:45, schrieb Jeffrey Walton:
I got a call from an overseas call center telling me about the
problems with the Windows machine I was using. They wanted to remote
in and fix things for me ... (Ignore the fact I use a MacBook Pro or
an ASUS laptop with Debian).
This is a common
Am 27.09.2014 17:28, schrieb d tbsky:
can someone give an example for the function? thanks for the help.
Not a programmer here, just grep -r'ed through the code, but maybe try
one of these:
G711A
G711_ALAW
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enforcement agency and then catch them when they pick up the money at
the Western Union counter.
I should write a book about that. :P
Cheers
Markus
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by executing sip show channel Call ID and
before figuring out the Channel by running core show channels concise,
but the issue is that the Call ID output from sip show channels is cut
off and limited to 16 characters.
Thanks!
Markus
Anna Crepes: Traubenzucker
+ Feldsalat spezielles Dressing (bringt selbst mit?)
Weitergeleitete Nachricht
Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.
Datum: Thu, 11 Dec 2014 15:34:39 +0100
Von: Markus unive...@truemetal.org
An: unive...@truemetal.org
Geschenke Moritz
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
=anotherpass
host=dynamic
nat=no
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
context=sipgate-priv
[sipgate-priv]
exten = _X.,1,NoOp
exten = _X.,n,Dial(SIP/${EXTEN}@sipgate-out)
Good luck,
Markus
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Am 13.06.2015 um 14:44 schrieb Luca Bertoncello:
If my Asterisk-configuration don't work, I don't have a phone and my wife
cannot work...
You will find out the day the switchover will be made by Telekom. If it
doesn't work, you'll analyze why and fix it within a few minutes, you
already did
Am 13.06.2015 um 13:54 schrieb Luca Bertoncello:
I think there are many german users in this ML, that use Asterisk with the
new line of Deutsche Telekom (Magenta Zuhause).
I don't think so. Most users will use the router provided by Telekom.
Anyway, after 15 seconds of Google'ing for Magenta
Am 04.06.2016 um 21:58 schrieb Steve Edwards:
1) If the caller ID matches '+493456789' (the first one), you goto the
'hangup' label. If it does not match, you goto the 'nohangup' label --
skipping the subsequent tests.
Doh! Removed :nohangup from every line but the last one and now it works
nothing here)
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+493456789"]?hangup:nohangup)
exten => _+X.,n(hangup),Hangup
exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" =
"anonymous"]?nocli:cli)
... more stuff that is handling the
to ignore me if it sounds like I'm suggesting you walk all the way
to the tool shed to fetch a chisel, when you know the screwdriver in your
drawer is already up to the job :)
you're right, it would be the better solution! But I'm simply too lazy
to implement that. :D
Regards
Markus
While we're at it, check out sngrep. Alex B. mentioned it on another
mailing list a couple days ago.
Screenshots: https://github.com/irontec/sngrep/wiki/Screenshots
Download: https://github.com/irontec/sngrep
Am 18.02.2017 um 05:10 schrieb Markus Weiler:
Hi Derek,
I think Homer (http
found:
"The dash (-) character is ignored in extensions and patterns except
when it is used in a pattern to specify a range in a character set. It
has no effect in matching or sorting extensions."
How do I do it right?
T
Am 28.10.2016 um 17:58 schrieb Max Grobecker:
why not using FILTER() in your dialplan to eleminate all chars that are not
numeric?
Like
Set(VAR=${FILTER(0-9+),${EXTEN}})
That would eleminate all characters you're not expecting.
That's great! Didn't know FILTER. Thanks!
--
There are inofficial RPMs for CentOS 7 available if you don't want to
mess with compiling: https://www.tucny.com/telephony/asterisk-rpms
Am 10.12.2016 um 15:47 schrieb christopher kamutumwa:
Hello support
am trying to install dahdi on centos 7 and am doing the make ommand and
below is result
Am 06.01.2017 um 12:07 schrieb Markus Weiler:
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.x
Can you try
exten => s,n,Set(CALLERID(num)=anonymous)
in your dialplan before passing the call to the PRI?
and/or
exten => s,n,Set(CALLERID(name)=anonymous)
and/or
exten => s,n,Set(CALLERID(all)=anonymous)
If it doesn't work, maybe replace the "anonymous" with "" or something else.
Sorry,
The task itself sounds like a job for an AGI script to me... check for
amount of calls, if 10, hangup one.
But how do you determine the priority of a call?
Am 07.11.2017 um 12:21 schrieb Jean Aunis:
Hello,
Has anyone already implemented some sort of call preemption in Asterisk
? I am
Bridge talks only to Moderator:
"Welcome to the moderator menu. The person who was talking in that very
second is on channel SIP/something-123456
Please press 1 to kick him out.
Please press 2 to ban him.
Please press 3 to return to the conference."
Something like that...
Thanks for
's my peer config:
[peer01]
host=1.2.3.4
type=peer
context=nowhere
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
timerb=2000
Thank you!
Markus
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=1.2.3.4
type=peer
context=nowhere
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
timert1=500
timerb=2000
timert1=500 is the default anyway, according to sip.conf comments...
Regards
Markus
Am 07.01.2019 um 17:23 schrieb Markus:
Dear list,
Asterisk 11.25.0 user here. I'm
I use ChanSpy for that.
This should get you on track:
https://community.asterisk.org/t/play-audio-file-on-channel-that-is-in-confbridge/67678
I don't use AMI, I just trigger asterisk binary through a shell script
(via AGI) like this to originate the call, it's easier for me:
ho "SET CALLERID \"\"<$CIDNAME>"
Then put your 10 CLIs in mobliecliall.txt.
PS: Shorter version for randomcli.sh, same functionality:
#!/bin/bash
CIDNAME=$(shuf -n 1 mobilecliall.txt)
echo "SET CALLERID \"\"<$CIDNAME>"
:-))
Regards
Markus
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