1: uniqueid=voip-1683277216.81923
level 1: linkedid=voip-1683277216.81923
level 1: sequence=64906
-- Streams --
Name: audio-0
Type: audio
State: sendrecv0
Group: -1
Formats: (alaw)
Metadata:
Thank you!
Markus
--
Am 17.10.2022 um 15:07 schrieb Joshua C. Colp:
The number is sent as the user portion in SIP. That can't have the "@"
in it, as that is invalid and against spec - it gets turned into "%40".
If what you ACTUALLY want is the user portion to be "anonymous" and the
domain portion to be
From: ;tag=as26264e65
My question is how can I get Asterisk to send a "@" instead of "%40"?
Thank you!
Markus
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Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
Hi All. We have old Asterisk servers, 1,89, (we cannot upgrade because
of several reasons) and we are now implementing SHAKEN via our
provider. We place a SIP call to our provider and they return a 302
(information below). I am trying to get
the world uses
"" as username/CLI so that it makes it into the "From: sip:@"
part. That call would also get rejected with 401 Unauthorized if I'm not
mistaken?
Is there a switch I'm missing?
Thank you, as always!
Markus
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Hi,
Am 08.06.2020 um 12:25 schrieb Antony Stone:
On Monday 08 June 2020 at 12:15:56, Markus wrote:
Hi list!
I'm getting this error frequently:
ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to
database server localhost: (2026) SSL connection error
running Linux on 2016-11-23 22:17:54 UTC
Database is a Galera cluster: galera-3-25.3.19-2.el7.x86_64
Asterisk is writing to the local MySQL instance and there's SSL between
the Galera cluster nodes.
Thanks!
Markus
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
-
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:
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
=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
'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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I received it :-)
Am 28.03.2018 um 22:44 schrieb Matt Fredrickson:
Just a test.
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Hi Group,
we're just wondering, in German we call the different types of phone-numbers
(Geographic,mobile,national,VoIP...)
Rufnummerngassen (phone number alleys ;-) )
Is there an english word for this?
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markus_wei...@mailworks.org
Hi All,
this seems to be a really neat module, that could really help us.
https://github.com/drivefast/asterisk-res_json
Any opinions about if we should use it in a production system?
Maybe from "official" asterisk side?
thanks
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
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,
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
Hi Derek,
I think Homer (http://sipcapture.org/) is the right answer :-)
HEP Agent will send the SIP trace to a remote Server (res_hep).
Markus
Am 18.02.2017 um 00:18 schrieb Tim Pozar:
You can tell it to just capture SIP traffic and not the RTP traffic.
Nice write up of using TCPdump
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
Nobody any idea?
It would be really helpful,
Markus
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
Stripped down example;
exten = 494X,n,Dial(SIP/4120089,15,w)
exten
the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded".
Is this a bug or a feature? :-) How could we handle this correctly?
SIP and Asterisk debug log below. Any help would b
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 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+),${EXTEN}})
That would eleminate all characters you're not expecting.
That's great! Didn't know FILTER. Thanks!
--
found:
"The dash (-) character is ignored in extensions and patterns except
when it is used in a pattern to specify a range in a character set. It
has no effect in matching or sorting extensions."
How do I do it right?
T
to ignore me if it sounds like I'm suggesting you walk all the way
to the tool shed to fetch a chisel, when you know the screwdriver in your
drawer is already up to the job :)
you're right, it would be the better solution! But I'm simply too lazy
to implement that. :D
Regards
Markus
Am 04.06.2016 um 21:58 schrieb Steve Edwards:
1) If the caller ID matches '+493456789' (the first one), you goto the
'hangup' label. If it does not match, you goto the 'nohangup' label --
skipping the subsequent tests.
Doh! Removed :nohangup from every line but the last one and now it works
nothing here)
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+493456789"]?hangup:nohangup)
exten => _+X.,n(hangup),Hangup
exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" =
"anonymous"]?nocli:cli)
... more stuff that is handling the
Am 20.08.2015 um 03:16 schrieb Pete Mundy:
Ah cr@p, sorry Steve, didn't mean to top-post there.
On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org
mailto:markus_wei...@mailworks.org wrote:
We started the 500 calls and used milliwatt app on the first and
record
. So you can get a good idea how the
quality is.
Call-Files are explained on
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Markus
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in the end was. But I remember rebooting was
important.
Markus
Am 11.08.2015 um 11:00 schrieb Stefan Viljoen:
Anybody else ran into this?
No, but I would ask myself why so many file descriptors are being used.
It sounds like you have a file descriptor leak (not being closed when
finished
the files.
Markus
Am 25.06.2015 um 18:09 schrieb Eric Cooper:
On Wed, Jun 24, 2015 at 08:15:06AM +0200, tux john wrote:
i would like to add email to fax functionality to the system. could someone
point me to the right direction to see how please?
I don't have a general solution, since I haven't
great,
would be the ideal time to comment the www.voip-info.org to contribute
to the community :-)
Markus
Am 14.06.2015 um 09:42 schrieb Luca Bertoncello:
Markus Weiler markus_wei...@mailworks.org schrieb:
Hi
from voipinfo...
If an Asterisk command specifies a sound file
will, if the language code is
de, first look for /var/lib/asterisk/sounds/*digits/de/*6.gsm before
falling back to /var/lib/asterisk/sounds/digits/6.gsm.
Markus
http://www.voip-info.org/wiki/view/Asterisk+multi-language
Am 14.06.2015 um 09:36 schrieb Luca Bertoncello:
Hi again
I'd like
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
definitely DNS...
check your Register lines...
Markus
Am 27.05.2015 um 20:14 schrieb Duncan Turnbull:
DNS failure could do this
Asterisk used to get stuck in a symmetric DNS request wait state which meant
everything ground to a halt as it waited for a reply while DNS timed out
Hi Patrick,
try voipmon, there it's free and you can even track MOS.
Markus
Am 25.03.2015 um 14:21 schrieb Patrick Beaumont:
Hi everyone.
We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load
.
What's the status with Asterisk REST API?
Any experiences on performance,stability,documentation, caveats? Any
toolkits for a fast start, Frameworks in any language? Hints? Best
practices?
Thanks for any insights!
Markus
Hi Nick,
maybe this will help?
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)
Markus
Am 27.02.2015 um 18:56 schrieb Nick Olsen:
Hello Everyone.
In my outbound
=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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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
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
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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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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--
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 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
very simple,
yet effective
http://www.palner.com/blog/171/asterisk-no-matching-peer-found-block/
Am 27.06.2014 16:58, schrieb Steven Howes:
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com
mailto:anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not
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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-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
feasible, and
switching to another country is not at option either. :)
All is good now!
Thanks!
Markus
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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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asterisk-users mailing
, 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 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
or tons of SIP servers.
cu, 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
; echo; echo; echo; sleep 1; done log.txt
Regards
Markus
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http
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
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
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
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
-with-dynamic-email-recipients/
HTH
Markus
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so much! :)
Regards
Markus
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asterisk-users
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
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
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
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
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
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Hi,
one more hint... (trying to translate the commands to english)
in Thunderbird open - Extras - Filter.. -
Filter-Name: enter Top Posting
Subject - Contains: enter Top Posting
Action: Delete
Markus
Am 02.01.2013 21:31, schrieb Steve Totaro:
On Wed, Jan 2, 2013 at 12:25 PM, j
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
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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Hi Felix,
ngrep -W byline port 5060|grep -B1 INVITE sip
Markus
Am 16.11.2012 17:50, schrieb Ruben Rögels:
Hi Felix,
you have several things to check:
netstat -a -n --udp --tcp
will show you connections and connection attempts on network layer level.
You have to look for incoming
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
hi,
try to catch in in a cron job per minute.
asterisk -rx 'module unload res_musiconhold.so'
Markus
Am 13.11.2012 19:15, schrieb Markus:
Am 13.11.2012 19:01, schrieb Eric Wieling:
module unload res_musiconhold.so
and
module load res_musiconhold.so
Great, that works, but only
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
just one originate action using
local channels.
Markus
Am 09.11.2012 11:38, schrieb Binan AL Halabi:
Hi,
Here is a starting point (WebRTC):
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
Regards.
// Binan
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 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)
--
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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.
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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asterisk-users
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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http://das-asterisk-buch.de/faxserver-mit-iaxmodem-und-hylafax.html
Regards
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
,Hangup()
Help! :)
Thank you,
Markus
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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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