Hi,
Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis:
> I am running Asterisk 13.30.0
> 40 core CPU (VM) VMware.
> CentOS 7
> 32 G ram
> 10G vmx network
>
> Should be plenty of room for anything...
>
> Yes asterisk is running 270% CPU...
> Is it not taking advantage of the 40
sts.digium.com>
> Subject: Re: [asterisk-users] Pickup with pjsip not working
>
> On Tue, Mar 1, 2022 at 7:16 AM Karsten Wemheuer wrote:
> > Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp:
> > > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer
> &g
Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp:
> On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer wrote:
> > Hi *,
> >
> > i am currently trying to migrate from chan_sip to pjsip. I am using
> > Asterisk version 18.10.
> >
> > In chan_si
Hi *,
i am currently trying to migrate from chan_sip to pjsip. I am using
Asterisk version 18.10.
In chan_sip information about the pickup was sent in the XML body of
the NOTIFY requests:
/---
\---
If I use pjsip, the pickup information is missing:
/---
\---
Many phones
Hi Luca,
Am Samstag, den 13.06.2020, 08:28 +0200 schrieb Luca Bertoncello:
> Hi!
>
> I have a Asterisk installation to manage my phones at home (provider
> is
> Deutsche Telekom).
> It works, but very often the voice is "broken"...
> Yesterday during a call it was very difficult to understand
Hi Stefan,
thanks a lot. It is working now.
Best regards,
Karsten
Am Freitag, den 01.05.2020, 18:40 +0200 schrieb Stefan Tichy:
> Hi Karsten,
>
>
> On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote:
> >
> > The server sends Server Hello,
Hi,
I have problems with SIP via TLS. Asterisk works as a client. The TCP
connection is established, followed by a client hello from Asterisk to
the server. The server sends Server Hello, Certificate, Server Key
Exchange and Server Hello Done.
Than Asterisk sends back a Alert (Level: Fatal,
Hi,
Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca:
> Hello
>
> Mynum: 6001 , Othernum: 6002.
>
>
> I can record as follows. But I do not enter individual records for
> each internal
> required. I want to do it more smoothly with a Macro.
>
> Thanks.
>
> exten => _6001,1,NoOp()
>
Hi,
I am currently evaluating asterisk 16. I have noticed an issue using
application playback. The beginning and the end of the audio file are
missing. If I use answer and wait(1) before playback, the beginning is
correct. I am using chan_sip, if this is of interest.
Best regards
Karsten
--
Hi Luca,
Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello:
> Hi list!
>
> My Problem: all calls to international numbers will be dropped after exactly
> 15 minutes...
> I have a VoIP-account by Deutsche Telekom.
> This is what I see when I call someone (my parents) and the
Hi Olivier,
Am Donnerstag, den 29.01.2015, 18:07 +0100 schrieb Olivier:
Hello,
I've just started to look at LDAP in IP telephony.
1. I've read parts of RFC2798 which defines inetOrgPerson class.
I could find homePhone or telephoneNumber (multi-valued) attributes
but nothing like
Hi,
Am Dienstag, den 16.12.2014, 16:32 +0100 schrieb Karsten Wemheuer:
Hi,
I got a weird behaviour in asterisk (original found in 1.8 but it is
still the same in 11.15.0). I have three phones communicating via
OpenSIPs with asterisk. Phone A dials 100 and asterisk calls
SIP/phone-b. Phone
Hi Neil,
Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry:
Doe anybody know of a way to redirect both channels from a bridge to
different dial plan extensions from the using the AMI.
Currently, as soon as I redirect one of the channels the other appears
to be dropped and gets
Hi,
I got a weird behaviour in asterisk (original found in 1.8 but it is
still the same in 11.15.0). I have three phones communicating via
OpenSIPs with asterisk. Phone A dials 100 and asterisk calls
SIP/phone-b. Phone B accepts the call. The User on Phone B places the
call on hold, dials 200
Hi Mike,
Am Montag, den 24.03.2014, 01:41 -0400 schrieb Mike Diehl:
Hi all,
I'm installing Hylafax on my Asterisk system. From what I've read, I
can either use IAXModem or T38Modem to provide the virtual fax device.
So at the risk of starting a religious war, which one should I use?
I don't
Hi Alex,
Am Dienstag, den 25.02.2014, 13:04 -0500 schrieb Alex Villacís Lasso:
El 25/02/14 08:30, Karsten Wemheuer escribió:
Hi Alex,
Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villacís Lasso:
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following
Hi Alex,
Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villacís Lasso:
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following
the setup guide at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb .
I want to run asterisk and kamailio on the
Hi,
Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb
gpxctawjc...@irational.org:
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
What Sipgate product are You using? At least in
Hi,
Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker:
Am 22.05.2013 16:39, schrieb Andrew Colin:
Hi guys,
Any idea why I am getting this error when someone tries to send me a T38
Fax?
Hi,
Maybe you have not allowed T.38 as acceptable codec ;-)
You can try
Hi,
Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus:
Am 07.05.2013 18:23, schrieb Sebastian Niehaus:
For some reason, t38modem tells hylafax the line is BUSY so there is no
fax send.
Well, I may add the log of t38modem (sorry for the ugly formatting)
Parts I consider
Hi,
Am Montag, den 04.02.2013, 14:45 +0100 schrieb Jonas Kellens:
Hello,
thanks you for your answer.
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone ?
as Kevin already written, set this in asterisk:
sendrpid=pai
trustrpid=yes
I don't
Hi list,
in asterisk 1.4 and maybe earlier it was possible to use voicemail
system with mailboxes starting with some special characters like *. The
line in voicemail.conf was like this:
*123 = , AB,,,tz=cet|attach=no|
Calling exten = s,n,Voicemail(*123,su) is working in asterisk 1.4.
In
Hi list,
Am Donnerstag, den 20.09.2012, 09:28 +0200 schrieb Karsten Wemheuer:
Hi list,
in asterisk 1.4 and maybe earlier it was possible to use voicemail
system with mailboxes starting with some special characters like *. The
line in voicemail.conf was like this:
*123 = , AB,,,tz
Hi Matthew,
Am Donnerstag, den 20.09.2012, 06:27 -0500 schrieb Matthew Jordan:
- Original Message -
From: Karsten Wemheuer k...@gmx.de
To: asterisk-users@lists.digium.com
Sent: Thursday, September 20, 2012 2:28:07 AM
Subject: [asterisk-users] Voicemail not working with vm boxes
Hii
Am Montag, den 11.06.2012, 16:12 -0700 schrieb motty.cruz:
Hello,
How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8
exten =
666,1,SIPAddHeader(Alert-Info:http://1.2.3.4/ringtones/ghost.wav)
exten = 666,n,Dial(SIP/10)
The above would not how to
Hi,
Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens:
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B
Hi,
Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi:
Hi
I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2)
I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI
Already installed).
according to
https://wiki.asterisk.org/wiki/display/AST/New+in+1.8
Hi,
while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I
want to notify a snom phone to reload its configuration. For this to
happen, I use the NOTIFY mechanism. I started the notify via AMI
command. Asterisk is bound to udp 25060, because all phones are
registered with a local
Hi,
a little extension to my previous post: The phone sends 200 OK for the
NOTIFY via proxy to asterisk, but asterisk seems to ignore this. About
500 ms later, the NOTIFY is repeated by asterisk. This continues up to
the final timeout (with the typical log message).
Karsten
--
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk
Hi,
Am Dienstag, den 14.02.2012, 11:32 -0600 schrieb Kevin P. Fleming:
This does appear to be a bug in Asterisk; please open an issue in JIRA,
and post the issue number here, so we can get someone looking at this
ASAP. Thanks!
Done, issue ASTERISK-19358. If I can do anything to test
Hi Bilal,
Am Sonntag, den 22.01.2012, 13:06 -0800 schrieb bilal ghayyad:
Hi All;
Is there a telephony card that contains analoge ports and E1s at the same
time?
Beronet in Germany produces modular media gateways as cards to plug in a
pc (PCI or PCI express) or as an external box. Each
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff on
the code it seems to me, that in version 1.8.7 there is an autoanswer in
application dial
Hi Richard,
Am Mittwoch, den 02.11.2011, 09:26 -0500 schrieb Richard Mudgett:
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff
Hi,
I try to setup a video call and I sometimes get no video.
I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same
LAN. Asterisk release is 1.8.7.0.
Call from Yealink to the Ninja is working fine, if I start the call in
video mode. In this case I can switch between voice-only
- if you start the call in voice-mode, the video codecs aren't
loaded.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten
Wemheuer
Sent: Wednesday, October 19, 2011 10:37 AM
To: asterisk-users
Hi,
Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot:
Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk
1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running
Linux on 2011-08-26 21:31:22 UTC]
The call flow is:
quintum gateway --
Hi Robert,
Am Donnerstag, den 25.08.2011, 13:28 -0400 schrieb Robert Huddleston:
https://issues.asterisk.org/jira/browse/ASTERISK-16981
Thank You for the link. I already found it a few hours later. I put some
debug output in the code and I think I found the location of the issue,
but I
Hi,
Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston:
Anyone else seen this?
I saw a jira but was in feedback status..
I just checked with a voicemail of 60 seconds. It was reported
in .txt-file with a duration of 19 seconds. So there is a bug. Do You
have a link to
Hi,
Am Montag, den 15.08.2011, 10:18 +1200 schrieb Alec Davis:
If you time the *8 just right so it is being handled during
the end of
the Dial then I got:
[Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ:
user_data
is NULL [Aug 11 16:26:18] ERROR[18458]:
Hi,
it seems to be fixed in 1.8.4. At least I can't reproduce it there.
Karsten
Am Mittwoch, den 15.06.2011, 09:29 -0400 schrieb Mike:
The same issue was present in 1.6 a few weeks ago and is fixed in
latest 1.6. Maybe latest 1.8.4 does not have this issue.
Mike
From:
Hi,
Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
$ sox foo.wav -e stat
just a remark for people using newer(?)/other version
Hi randall,
Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
of span 2
Your telco provider has crc on or off , that is not matching
Hi Felix,
Am Mittwoch, den 16.02.2011, 12:47 +0100 schrieb Felix Dong:
Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP
connection. Can anybody help?
thanks a lot.
are You trying to echo between local phones or is it a external call via
some VoIP Provider?
In
Hi,
Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman:
Replys from Bryant
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com
wrote:
I am having issues with Blind Transfer on asterisk 1.8
What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
Am Donnerstag, den 21.10.2010, 16:27 +0200 schrieb Karsten Wemheuer:
Hi,
I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.
The following
Hi,
I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.
The following is working:
User A calls user B, B accepts the call, user A than transfers to user C
Hi,
Am Mittwoch, den 20.10.2010, 01:54 -0200 schrieb Flavio Miranda:
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Just one more question, what it means the RED under alarms when I
type dahdi show status. It should be OK?
the RED-alarm usually
Am Samstag, den 16.10.2010, 14:00 -0500 schrieb Shaun Ruffell:
On 10/16/10 12:47 PM, Karsten Wemheuer wrote:
Hi,
Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell:
On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux
Hi,
Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell:
On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
older pc system with Celeron CPU
Hi,
I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
card. The system starts without any errors.
I discovered a severe issue. The kernel
Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer:
Hi,
while testing current release candidate 1.8.0-rc2 I stumbled on a weird
behavior. I did not find any hints in the archives or at the bug
tracker.
Two SIP-Clients are connected (both on the local net, no NAT). The RTP
Hi,
while testing current release candidate 1.8.0-rc2 I stumbled on a weird
behavior. I did not find any hints in the archives or at the bug
tracker.
Two SIP-Clients are connected (both on the local net, no NAT). The RTP
stream flows directly between the phones. If I set phone A on hold, the
Hi,
Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades:
Olivier wrote:
Hello,
I've got a running system in which logs are full of messages such as:
[Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 2
The strange
Hi Olivier,
Am Freitag, den 11.06.2010, 14:27 +0200 schrieb Olivier:
2010/6/11 Karsten Wemheuer k...@gmx.de
Hi,
Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades:
Olivier wrote:
Hello,
I've got a running
Hi,
Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler:
On Tue, 16 Feb 2010, Armin Schindler wrote:
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.
Oh thanks, I
Hi,
Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude:
Hi,
I'm having trouble getting feature codes to work in Asterisk 1.4.21.2.
Features.conf contians this:
blindxfer=##
atxfer=*2
automon=*1
disconnect=**
I'm really most interested in getting disconnect to work so that
Hi,
Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune:
Hi,
I tried again getting DTMF detection on my ISDN devices with dahdi going
again. I used the channel debug to see whether asterisk sees the frames
and detects them as DTMF.
Interestingly here's what works:
1.
Hi Bart,
Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher:
I don't understand this message:
[2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos:
Unable to set TOS to 184
You did not tell us, which version of asterisk You are running.
The kernel restricts setting
Hi,
Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton Sapale:
That's the strange thing. Nothing shows when monitoring the service
in debug. On the phone, however, I do see a connection time-out
error. I guess this might indicate that the device is attempting to
connect to the service
Hi Michael,
Am Samstag, den 13.12.2008, 23:09 +1300 schrieb Michael:
For some odd reason the call registration issue doesn't seem to stop it
working, except a few seconds after Hylafax answers the call it hangs up, I
suspect because Asterisk only supports T38 pass through.
Here is my data
Hi,
Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie:
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?
The SNOM 300
Hi,
Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the
Hi Eric,
Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort:
resolve.conf and dns is working. The problem persists.
/var/log/asterisk/messages shows a few notices and warnings on
res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading
of these in modules.conf asterisk
Hi Ruddy,
Am Freitag, den 12.09.2008, 21:38 -0400 schrieb Ruddy Gbaguidi:
But user just needs to enter * instead of *#
We are doing this because 80% of the callers already have an account,
so, instead of playing :
If you have an account, press 1, if not press 2
we prefer to play
Enter
Hi,
Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi:
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
Hi Ruddy,
Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi:
Thanks for the hint. Sorry about that.
If I use your soution, I cannot make any difference between a user
pressing * and a user that reach the timeout because he didn't enter any
digit.
In both cases, I will have an
Hi,
Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène:
My iax.conf looks like this:
[iaxmodem]
type=friend
host=127.0.0.1
secret=x
context=fax-out
permit=127.0.0.1
disallow=all
allow=ulaw
after editing inittab I reload it by running: /sbin/init q
I also
Hi David,
Am Sonntag, den 20.07.2008, 11:57 +0200 schrieb David Ashwood:
Morning guys and gals,
I’d like to be able to run some code when a device (soft/hardphone)
authenticates to Asterisk.
I remember reading somewhere that there’s the possibility of part of a
dialplan can be run
Hi,
Am Mittwoch, den 25.06.2008, 08:42 -0400 schrieb Jerry Geis:
I am running asterisk from svn check out from yesterday Jun 24.
I started with 1.4.20, then 1.4.21 then svn.
I am getting:
pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed
segment fault.
I am
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time.
Hi,
Am Freitag, den 04.04.2008, 13:03 + schrieb Tony Mountifield:
In article [EMAIL PROTECTED],
Faraz R. Khan [EMAIL PROTECTED] wrote:
Thinking out loud: write a asterisk call file (when the calling user
presses 5) which keeps on trying to connect the two.
I thought about that, but
Hi Martin,
Am Freitag, den 28.03.2008, 14:27 +0100 schrieb martin f krafft:
[...]
So calls are going via an asterisk bridge and the symptoms of my
problem are:
1 if C450IP calls softphone, they can talk fine
2 if softphone calls C450IP, voice only goes from C450IP to
softphone, not
Hi Paul,
Am Dienstag, den 05.02.2008, 10:10 +1100 schrieb Paul Hales:
With some of the phones (snom, for example) you can turn off mwi, so the
phone will only accept one call at a time. Much easier.
PaulH
Thanks for Your answer. Unfortunaly turning call waiting off is not an
option for me.
Hi Stefan,
Am Dienstag, den 05.02.2008, 10:30 +0100 schrieb Stefan Guenther:
Hi,
according to the description of Pickup() on page
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
I can use this command to pickup a call at a certain extensions.
When I try this with e.g.
exten
Hi,
I want to use GROUP_COUNT to limit calls to a specific destination. From
somewhere on the wiki I am using the following context:
exten = 200,1,Set(GROUP()=${CALLERID(num)})
exten = 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} = 1]?BLOCK)
exten = 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten =
Hi,
Am Donnerstag, den 06.12.2007, 11:30 -0500 schrieb Jon Pounder:
Quoting Ricardo Carvalho [EMAIL PROTECTED]:
I only see one explanation to my problem...
GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and
the power injector does only PoE mode B of the IEEE
Hi Ray,
Am Dienstag, den 09.10.2007, 10:10 -0500 schrieb Ray Chen:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to get
the ingress and egress sip call id for a call
even need to use ODBC, because Asterisk has native
support for sqlite.
Are You shure the native support of asterisk is for SQLite3 as the
original poster asks for? AFAIK * supports SQlite (Version 2, not 3),
which has a completely different API.
Karsten Wemheuer
Hi,
Am Samstag, den 08.09.2007, 09:44 + schrieb wassim darwish:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04
-0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On
Friday 07 September
Hi,
Am Donnerstag, den 05.07.2007, 14:58 -0400 schrieb Andrew Kohlsmith:
On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote:
Already did that. I use ASSP for filtering. Digium and associated
mailing lists are white listed. There was only 1 attempt for deliver
and there were no delays.
Hi Morgan,
Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy:
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729
On Fri, Apr 06, 2007 Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've
On 03/22/2007, Torge Szczepanek wrote:
Hello list!
I have a Asterisk 1.2.10 running using the package from Backports.org
for Debian Sarge.
I have setup chan_capi (0.6.5 from Backports) and it seems that I am
only able to use on B-Channel.
When trying to place the second call I get:
Hello,
Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon:
1. The smallest mini-ITX case I found that accepts a PCI card is the
Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know
if it fits? I didn't find its width, and apparently, the C138 will not
Hi,
maybe I am a little bit late with this answer.
I take a look at Your config and the debug output.
snip
Provider --te11xp--- asterisk ---te11xp-- nortel merridian
option 11c
snip
zapata.conf
-
context=from-pstn
switchtype=dms100
Hi,
On Fryday, 2007-02-02 François Delawarde wrote :
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes through it?
François.
On a SIP-to-SIP-Call Asterisk is not neccessarily in the voice flow,
so this does not
Hello,
AFAIK the feature CD (call deflection) is only possible on
point-to-multipoint links, is this correct?
I've heard about the feature partial rerouting which should do the
same on point-to-point-links. Is this implemented in either bristuff or
chan-capi(-cm)?
Thanks in advance,
Karsten
Hello,
while testing the following scenario, I ran into trouble:
One * box with two AVM active controllers in Point-to-Point-Mode is
connected to another * box with ZapHFC/Quad-BRI cards using bristuff in
NT-mode.
All is working fine, I can call from one box to the other and vice
versa.
But if
Hello,
On Mo, 27 Feb 2006, Armin Schindler wrote:
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
In detail:
When all lines are connected, the first two calls are placed on line 1
(which is on controller 1). The next two calls are placed on line 2 (on
controller 2)
If I'll cut line 2
Hello Armin,
Am Mo, den 27.02.2006 schrieb Armin Schindler um 20:23:
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
Hello,
On Mo, 27 Feb 2006, Armin Schindler wrote:
This is not a bug, just normal behaviour.
chan_capi does not know about the status of the ISDN line, it assumes
Hi,
I answer to my own posting...
On Sun, Jan 15 2006 Karsten Wemheuer wrote:
Hi,
I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When
I activate music-on-hold on a SIP-to-SIP connection, the music sounds
like in a fast-forward play mode. On the *-console I can see
Hi,
On Mon, Jan 16 2006 Louis-David Mitterrand wrote:
Hello,
Using asterisk-1.2.1 I am trying to convert my music-on-hold files from
.wav to alaw:
% sox moh.wav -r 8000 -c 1 moh.al resample -ql
The file sounds fine when listened with:
% sox mox.al -t ossdsp /dev/dsp
Hi,
I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When
I activate music-on-hold on a SIP-to-SIP connection, the music sounds
like in a fast-forward play mode. On the *-console I can see much lines
like this:
-- Silence suppression is disabled (option_silence_suppression=0
Hello Armin,
On Mo, 2 Jan 2006 Armin Schindler wrote:
I don't think it is necessary to exclude it. Just build chan_capi-cm and
overwrite chan_capi.so as well as remove the app_capi* modules from your
installation.
Armin
Many thanks, it is working.
Karsten
Hello,
as You are running two processes handling SIP (asterisk and openser), I
think the Call-File addresses the wrong instance.
If Your callfile contains a line like
Channel: SIP/accountname
try something like
Channel: SIP/[EMAIL PROTECTED]:port
where ipaddress and port
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem.
I want to signal busy to an incoming call, but that doesn't work.
The dialplan looks like this:
exten =
Hello Armin,
On Mo, 02.01.2006 Armin Schindler wrote:
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
Hello,
first of all, I say Happy New Year to this list!
While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem
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