check linphone. AFAIK there is a linphonec for command line. I think
there is also a joshua (sample application) for osip which is CLI based.
regards,
klaus
Olle E. Johansson wrote:
Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
Take a look at http://ebills.sourceforge.net/
I uses latex to create nice pdfs.
regards,
klaus
Christopher Snell wrote:
On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon
[EMAIL PROTECTED] wrote:
What I would like to know is has anyone found an open-source billing
platform that performs basic billing
Hi all!
I've sucessfully bridged an 3G-H324M video call through asterisk using
ISDN PRI zap channels. Now, I want to monitor the call and dump it into
a file for later analysis of the H324 call setup.
I tried to use the Monitor() application, but I breakes the session -
looks like it does some
Hi Stefan!
1. You also have to modify the RTP channel, as floor control will be
done using RTCP.
2. The biggest problem is the lack of compatible hand sets. There are
several standards (Nokia, Ercisson, OMA..) and it is hard to find
working handset.
regards,
Klaus
Stefan Gofferje wrote:
Hi
try SIPPS from Ahead (Nero)
klaus
Adi Linden wrote:
I am looking for a German language softphone. Is there such a thing?
Adi
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To
Hi!
I'm using a PRI card. When a call arrives, I want to answer the call and
dump the binary data received on the B-channel into a file or stdout or
to the console (for debugging the B-Channels).
Is this possible?
regards,
klaus
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There is a thread in asterisk-dev, e.g.
http://lists.digium.com/pipermail/asterisk-dev/2005-January/008761.html
regards,
kalus
Kuniyoshi Murata wrote:
Hi,
Is there any future possibility that Asterisk will be compatible with
connection to 3G video mobile phone such as Nokia 7600, Nokia 6630
Date: Tue, 1 Jul 2003 14:37:20 +1000
From: Andrew Radke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] A solution for SIP and NAT
...
So I've started a really simple SIP and RTP proxy project, SaRP, on
sourceforge.net. Yesterday we uploaded 0.2 of the perl based release.
Correct, MSN Messenger 5.0 does not support SIP. But the upcoming
Windows Messenger 5.0 will do. I have installed the beta version from:
http://msnarea.ezonate.co.uk/?act=downloadsCODE=01cat=1
I think, the new version supports more codecs.
regards,
Klaus
Message: 3
To: [EMAIL PROTECTED]
http://developer.berlios.de/projects/mystun/ works fine for me. Tested
on debian sarge with one NIC and two IP addresses for this NIC.
Klaus
AJ Grinnell wrote:
What is the best free stun server out there? The one that I have looked at
from vovida requires two NICs. Is this neccessary?
You could try the following:
If ser detects an outgoing call (like hisdomain.com), it starts an
external script (cpl-c module in ser) which lookups the domain (dig,
nslookup) and verifies it against the IP address of the asterisk box.
You would also have to take care of multiple returned IP
Hi!
I had the same problem yesterday - but google revealed:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044867.html
furthermore the app_dtmftotext application fails to start, so i deleted
it - I hope this has no influence :-(
regards,
klaus
Terry Goodwin wrote:
Thanks for
Brian D'Arcy wrote:
ast_load_resource: libspandsp.so.0: cannot open shared object file: No
such file or directory
I copied the libspan* files from /usr/local/lib to /usr/lib and then
asterisk started!
klaus
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Jeremy Hall wrote:
If by authentication by mobile number you mean the caller ID received,
that is not secure at all. CallerID is very easy to spoof when you have
a digital line (certain types, of course.) For example, when I call out
from my Asterisk box, if I prefix the number with 9, it sends
, don't come hungry! Nevertheless
liquid food (drinks) is available.
We meet in the library (in the back of metalab): http://metalab.at/wiki/Lage
For people who do not know me - I will wear my openser T-Shirt (unless
it is in the laundry).
regards
Klaus
Klaus Darilion schrieb:
Hi!
I proudly
If you do not like giving your SIP credentials to others, you can
install a SIP phone like http://www.pernau.at/kd/voip/ActXPhone/ easily
on your own homepage. It does not allow registration at the SIP proxy,
but this can be added very easily (visual basic).
regards
klaus
kevin ling wrote:
Ryan Amos wrote:
I use a PE2850 with CentOS 4.2 on it (as parent says, it is essentially
RHEL 4 without the support contract.) Extremely stable; no problems with
asterisk at all. Dell makes 2 PCI riser cards for this server, I believe
one of them has 5v slots. I have a 3.3v card so I can't tell
Hi!
Are there any multiport ISDN interface cards (PRI and BRI) which support
pass-through in power-off mode. (I want to use Asterisk between the
telco line and the existing PBX and I want pass-through when the power
of the Asterisk server is switched off).
regards
klaus
siptapi
Kashif Naeem schrieb:
Hello All,
Can anyone please recommend me some good Click 2 Dial application ? We
need to dial using Microsoft Outlook Business Contact Manager.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Have you seen the footnote on http://sip.fontventa.com/content/view/26/53/ ?
klaus
pradeep bhimellu schrieb:
Hello there,
I have just finished the Asterisk setup for 3G video
calls and tried to test with my Samsung SGH-G800 but
no success.The phone says Dialing for 20-30 seconds
and call is
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available (except on the mailing list and
the voip-info wiki (which is usually very old))?
I think this is not possible. If you take a look at main/rtp.c there is
no config option for an IP address.
regards
klaus
Jun Yin schrieb:
some vendors(like alcatel-lucent) developed a kind of sip proxy which
includes two parts: one sip signaling module and one or more voice
modules. voice
Klaus Darilion schrieb:
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Answer myself: I have found the documentation about chan_local's options
in doc/tex
Johansson Olle E schrieb:
26 jun 2008 kl. 10.17 skrev Klaus Darilion:
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).
Another reason could be that the telco blocks video calls.
Grygoriy Dobrovolskyy schrieb:
You have 2 choices to pickup someone's phone with snom's
1: imagine yourself prefix of pickup, let's say 4
exten=4XX,1,Pickup([EMAIL PROTECTED])
so if u call 4 + phone number you will pickup that one.
Second you can add pickupgroup=number for each phone
you also need (as stated in the bug report) the patch
10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch from
http://bugs.digium.com/view.php?id=10217
This enables LCC in chan_zap. Is this was done some time ago I do not
remember anymore who it is activated, I think you have to add the
Hi!
I want to test Asterisk--Siemens HiCom integration using Q.SIG instead
of ISDN. I did not find any documentation about Asterisk und Q.SIG.
Thus, I wonder is it sufficient to set switchtype from euroisdn to
qsig or are there any other things which I have to take care of?
Thanks
Klaus
Hi!
I wonder how Asterisk measures the call duration. The CDR files have a
accuracy of seconds. Thus, what happens if the call duration is 0.3
seconds. What will Asterisk report? 0 seconds? 1 second?
What logic will be used by Asterisk: floor? ceil? round?
thanks
klaus
Hi!
Is there a way in Asterisk to get intermediate accounting records? E.g
Cisco gateways send start, stop and and regular interval intermediate
accounting records.
For example if there is a call and Asterisk (or the hardware) crashes
during the call we do not have a CDR for this call, not
Steve Murphy wrote:
On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote:
Hi!
I wonder how Asterisk measures the call duration. The CDR files have a
accuracy of seconds. Thus, what happens if the call duration is 0.3
seconds. What will Asterisk report? 0 seconds? 1 second?
What logic
Use the ENUMLOOKUP function, e.g.:
[from_sip_phone]
exten = _00X.,1,Set(enumresult=${ENUMLOOKUP(+${EXTEN:2})})
exten = _00X.,2,GotoIf($[${enumresult} = ]?103:3)
exten = _00X.,3,Dial(SIP/${enumresult},90)
exten = _00X.,4,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:5)
exten =
Hi!
Does somebody know a Thunderbird extension to playback the voicemail
(wav attachment) directly in Thunderbird? (or another neat workaround?)
thank
klaus
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Hi!
Using setvar in a peer configuration (sip.conf) I can set the channel
variables for the incoming channel. Is there a similar method which
allows me to load these variables also for outgoing channels (e.g. to
load callee preferences)?
thanks
klaus
Hi!
Is it possible to specify the timezone in the GotoIfTime application?
E.g. I want to route the call if it is 9:00-10:00 in Austria/Vienna or
10:00 - 11:00 in New York.
This is needed for example if the time based routing for the office in
New York is done on an Asterisk server running
I answer myself: since Asterisk 1.6 you can use the SIPPEER function to
retrieve the peer's setvar variables.
regards
klaus
Klaus Darilion schrieb:
Hi!
Using setvar in a peer configuration (sip.conf) I can set the channel
variables for the incoming channel. Is there a similar method which
Hi!
When I enter the Asterisk console and press the up key I get the
command history. But the quit is twice in the history. Why?
thanks
klaus
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Hi!
i have a problem with the PICKUPMARK of the Pickup() application.
E.g. A calls B. B is ringing. C wants to pickup B.
To make this work with PICKUPMARK I have to add the variable PICKUPMARK
to B. But how can I do this? B is just created inside the Dial()
application.
thanks
klaus
PS:
: 1111 10100100
But Asterisk only uses one byte for the ä.
Is there a way to configure Asterisk to encode the ä as UTF-8?
thanks
klaus
--
Klaus Darilion
nic.at
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Benny Amorsen wrote:
KD == Klaus Darilion [EMAIL PROTECTED] writes:
KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I
KD have a displayname with special characters?
KD E.g. if I want to have the Umlaut ä in the display name:
KD callerid=Jeff Gräser 11
Is your sip.conf UTF-8
Klaus Darilion wrote:
Benny Amorsen wrote:
KD == Klaus Darilion [EMAIL PROTECTED] writes:
KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I
KD have a displayname with special characters?
KD E.g. if I want to have the Umlaut ä in the display name:
KD callerid=Jeff Gräser 11
Hi!
Is it possible to deactivate RTCP? (I am using 1.6)
thanks
klaus
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Hi!
For T.38 Asterisk uses the port defined in udptl.conf. Is there a
workaround (I am using 1.6) for using the same port as RTP also for UDPTL?
regards
klaus
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Hi!
Is there somewhere a statement from Digium how long they will support
Asterisk 1.4?
thanks
klaus
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Hi!
What is the preferred way to make database lookups from within the dialplan?
I only know the MYSQL function from asterisk-addons. Are the other
methods too? (e.g. for postgresql, unixodbc)
thanks
klaus
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Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
The only thing were nat=yes is bad is if you have an asymmetric client.
I do not know any
Alex Balashov schrieb:
Klaus Darilion wrote:
Actually I would nat=yes always, even if clients are not behind NAT os
otherwise the clietn can put some garbage into the contact header (e.g.
IP address of an upstream provider) and influence routing.
No. There is a specific reason RFC
Alex Balashov schrieb:
Steve Totaro wrote:
Alex is going to cling to to the RFC as if it were the gospel, and not
look at what would essentially be a good thing.
The RFC is not the gospel, but nor is it just a request for comment,
historical nomenclature aside.
It is the de facto
this yourself?
Do you know how it compares to MYSQL function and func_odbc?
regards
klaus
regards,
Wolfgang
Klaus Darilion schrieb:
Hi!
What is the preferred way to make database lookups from within the dialplan?
I only know the MYSQL function from asterisk-addons. Are the other
methods too
(caller preferences, LNP,
LCR...)
regards
klaus
Jared Smith schrieb:
On Thu, 2008-11-13 at 15:16 +0100, Klaus Darilion wrote:
What is the preferred way to make database lookups from within the dialplan?
The preferred method is to use func_odbc, which takes SQL queries and
builds custom
Alex Balashov schrieb:
Klaus Darilion wrote:
This is a different scenario. In this case of course I want the public
IP of the client, not of the load balancer. So, yes - in this case
nat=no is useful for Asterisk. Nevertheless I ignore the IP provided by
the client in the contact
Alex Balashov schrieb:
Klaus Darilion wrote:
Of course we know that we should implement RFC conform. But RFC 3261 has
ignored the fact that the Internet is full of NATs and standard conform
implementations can not work. This in the case of SIP it necessary to
break the RFC
Hi!
I wonder why users.conf generates a SIP user and a SIP peer? Why is it
not possible to set type=... in users.conf? (Asterisk 1.4.22)
thanks
klaus
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Hi!
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine,
Thanks Dave, Philipp and Richard!
klaus
Richard Lyman wrote:
Philipp Kempgen wrote:
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what
since 1.4 you can also use
setvar=foo=bar
in sip.conf when configuring the peer. Then the channel variable foo is
automatically set to bar for calls initiated by this peer.
regards
klaus
Philipp Kempgen wrote:
Grey Man schrieb:
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady
There is also type=[user|peer|friend] in chan_iax and chan_h323
there is also type=h323|alias in chan_h323
maybe it is better to use in users.conf another variable, e.g.
siptype=
or
h323type=
regards
klaus
Tzafrir Cohen schrieb:
On Tue, Dec 23, 2008 at 10:35:19AM +0100, Klaus
http://bugs.digium.com/view.php?id=14188
regards
klaus
Klaus Darilion schrieb:
There is also type=[user|peer|friend] in chan_iax and chan_h323
there is also type=h323|alias in chan_h323
maybe it is better to use in users.conf another variable, e.g.
siptype=
or
h323type
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
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Grygoriy Dobrovolskyy schrieb:
core show function SIPPEER
Does not work. Using the SIPPEER function you have to know the name of
the peer already.
regards
klaus
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
since 1.4 you can also use
Hi!
Currently I provision user account in users.conf. But I do not like that
VoiceMail writes to users.conf when the voicemail password is changed.
Is there a possibility to store the vmsecret in another place? (another
file or DB)?
thanks
klaus
Hi!
All the AEL examples have a semicolon after the closing curly bracket, e.g:
context test {
1 = Hangup();
};
but without ; it works fine too, e.g:
context test {
1 = Hangup();
}
So - what is the reason for the ; after the closing curly bracket?
thanks
klaus
Hi!
I wonder how to configure a SIP peer which
- requires authentication for calls to sent to the peer
- needs to authenticate for incoming calls
I want to have different username/password for incoming and outgoing
direction.
Thanks
Klaus
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on realm
Klaus Darilion wrote:
Hi!
I wonder how to configure a SIP peer which
- requires authentication for calls to sent to the peer
- needs to authenticate for incoming calls
I want to have different username/password for incoming and outgoing
direction.
Thanks
Klaus
Watkins, Bradley schrieb:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Klaus Darilion
Sent: Thursday, January 08, 2009 8:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=YES
Of
Hi!
I have asterisk 1.4.22 configured to register to a SIP proxy. The
problem is that if for some reason the registration fails once (e.g.
REGISTER-401-REGISTER-403), Asterisk does not try to reregister again -
I have to sip reload to trigger a reREGISTER.
I tried the default SIP settings and
Hi!
I use unixodbc to connect to mysql. When the connection to the DB is
lost (e.g. restarting the Mysql server) the reconnect takes 5 seconds:
[Jan 9 13:22:15] WARNING[13899]: res_odbc.c:113
ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
reconnect...
[Jan 9 13:22:20]
Klaus Darilion schrieb:
Hi!
I use unixodbc to connect to mysql. When the connection to the DB is
lost (e.g. restarting the Mysql server) the reconnect takes 5 seconds:
[Jan 9 13:22:15] WARNING[13899]: res_odbc.c:113
ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting
Hi!
I want to detect brute-force password hacking attacks - thus if there
are too many failed login attempts for a SIP account I want to lock
this account.
Does somebody have any ideas how this could be implemented?
thanks
klaus
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Hi Philipp!
thanks for the detailed explanation.
Philipp Kempgen schrieb:
=== Amooma ===
* http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
These files are generated by our web-based text-to-speech engine.
Pros: If you need additional custom prompts, just go to
Dave Platt schrieb:
Bad plan? Could quite easily turn into a DoS.
If the reaction is to lock the account, I agree, it might
leave you prone to a denial-of-service attack.
A better way would be to use iptables to start dropping
packets from the IP address(es) involved in the attack...
I use Asterisk 1.4.22 without problems:
Contact: sip:+43720123456...@1.2.3.4:5160
regards
klaus
mailinglists schrieb:
Hi all!
Am I missing some configuration or is it simply a bug: If
Asterisk chan_sip is configured with bindport=5062, the port is missing
on the outgoing SIP messages
open a bug report
mailinglists schrieb:
Hi all!
Am I missing some configuration or is it simply a bug: If
Asterisk chan_sip is configured with bindport=5062, the port is missing
on the outgoing SIP messages contact header.
This resulting in in-dialog messages sent to port 5060 ... where
Philipp Kempgen schrieb:
=== Amooma ===
* http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
These files are generated by our web-based text-to-speech engine.
Pros: If you need additional custom prompts, just go to
http://www.amooma.de/tts/ and generate them and the voice will
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g.
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP softswitch.
This is IMO a stupid limitation. There are dozens
Joshua Colp schrieb:
- Klaus Darilion klaus.mailingli...@pernau.at wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside
the dialplan?
No. Part of the reasoning is that Asterisk is meant to be a
multi- protocol PBX
IIRC FaxGateway is intelligent and works in both directions.
What are the problems?
klaus
Alex Balashov schrieb:
Well, T.38 works over IP, not TDM...
James Lamanna wrote:
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem
Just google for your subject.
short: insmod just tries to load one module. modprobe checks
dependencies and loads needed kernel modules too.
klaus
Olivier schrieb:
hello,
Here (http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html) you
can read :
cd qozap
modprobe zaptel
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant
Johansson Olle E schrieb:
15 jan 2009 kl. 12.42 skrev Klaus Darilion:
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside
/asterisk/causes.h is a good reference for now.
/O
Thanks.
on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote
14 jan 2009 kl. 14.02 skrev Klaus Darilion:
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards
I do not know cordiip thus I do not know how these 3 different accounts
are signaled to you, but some tips:
A SIP peer is always identified by host:port - thus there is at peer
level no way to differ them. But in the register command you specify the
contact to be called, e.g. 1646H25.
Hi!
If a user has recorded a new voicemail message (e.g. unavailable
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message
Isn't it possible to cancel the recording?
thanks
klaus
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
If a user has recorded a new voicemail message (e.g. unavailable
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message
Isn't it possible to cancel the recording
What you need is a so called T38Gateway application.
there is a patch o the tracker which you might want to try:
http://bugs.digium.com/view.php?id=13405
klaus
Steve Gladden schrieb:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP
Hi!
I have two identical SIP accounts on Asterisk 1.4.22. One account is
registered with eyebeam, the other one is registered with a SNOM phone.
When using the eyebeam client DMTF detection works fine, when using the
SNOM phone many digits are missing in the DTMF detection.
I analyzed with
Alex Balashov schrieb:
How are you testing DTMF detection with the Snom UA?
The Voicemail(u...@context) application asks the user for the voicemail
password.
Using eyebeam everyhing works fine. Using SNOM (320 FW 7.3.10a) it works
almost never.
regards
klaus
Klaus Darilion wrote:
Hi
Hi!
I have the following scenario:
Asterisk
INVITE- |
--200,ACK-- |
Playback(Foo)
|
Dial(..)
| -INVITE-
| -404. ACK--
|
As my extension configuration stops after the Dial command I
features.conf for transfers
for call forwardin you need some application logic.
e.g.
_**21**. = {
Set(NUM=${EXTEN:6}); // contains the new target
// now store this number somewhere, e.g. astdb, odbc ...
...
}
context fromPstn {
1234 = {
// check if user has actived forwarding
Jared Smith schrieb:
On Tue, 2009-01-20 at 16:08 +0100, Klaus Darilion wrote:
As my extension configuration stops after the Dial command I expect
Asterisk to hang up the call. Instead I see on the console:
|
|
== Auto fallthrough, channel 'SIP/81.16.153.183
Sean Bright schrieb:
Klaus Darilion wrote:
Ok.
Just for the info to others: the 10 seconds are hardcoded in pbx.c
What line in which version?
at least in 1.4.22:
grep -r -A 10 -B 10 Auto fallthrough *
klaus
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on the receiving side for setting
the hangupcause instead of doing SIP-hangupcause mapping ?
Then the original hangupcause is not lost.
regards
klaus
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb
Hi!
After heavy debugging I found out that I had packet loss from the SNOM
phone to Asterisk. Connecting the SNOM phone from the office switch
(with autodetect) to an HP Procurve (with autodetect) solved the issue -
packet loss is gone and DTMF works.
regards
klaus
Klaus Darilion schrieb
Hi!
I want to configure voicemail to send emails with the date of the
message in German/Austria, that means:
Montag, 26 Jänner 2009 instead of Monday, 26 January 2009
voicemail.conf refers to man strftime. This refers to the current locales.
So, I tried
export LANG=de
export
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
I want to configure voicemail to send emails with the date of the
message in German/Austria, that means:
Montag, 26 Jänner 2009 instead of Monday, 26 January 2009
voicemail.conf refers to man strftime. This refers to the current locales
Hi!
Is it possible to configure a negative TTL (number was not found in
Dundi) for DUNDI?
regards
klaus
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Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I
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