ChallengeSent? That should occur also on
legitimate login processes...
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
, sip:goip) in new stack
-- Executing [+3584172x@messages:4]
Hangup(Message/ast_msg_queue, ) in new stack
== Spawn extension (messages, +3584172x, 4) exited non-zero on
'Message/ast_msg_queue'
Is there a way to make * ignore the comma in the string here?
-S
--
(o_ Stefan Gofferje
?
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
On 06/03/2014 01:53 PM, Patrick Laimbock wrote:
Have you looked at Call Completion Supplementary Services (CCSS)?
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096
PSTN doesn't support that here.
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User
On 06/03/2014 06:06 PM, Eric Wieling wrote:
Have you tried RetryDial()?
I want it to be a conscious decision and not just automatically in every
call. For the vast majority of my call I can just try some time later
but some people I need to get a hold of ASAP sometimes.
-S
--
(o_ Stefan
On 06/03/2014 12:44 PM, Israel Gottlieb wrote:
you could save the info in astdb for the last call per extension and
then pull it from there
I guess I'll have to do this then :).
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_
and when
it's busy again, you have to again hang up and press the button again...
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic
to once a day or so.
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
start monitoring proactively because being in the
top 20 of attacker-IPs ain't good for their reputation...
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
-A FORWARD -s ${ADDRESS} -d $ANY -j LOG --log-prefix
Packet log: COUNTRY DROP
done
done
echo Done. Started: ${DATE}, finished: $(date)
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click
? Or is the project dead?
Or is nobody tinkering any more and everybody buying some turnkey-stuff?
Just wondering...
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
is the last bit that my * is missing before being the
perfect PBX, so I hope, somebody here could help me with that :).
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
don't favor GSM-PCI-cards because I'm just building a new asterisk
based an an Atom board in a small casing.
--Stefan
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
out http://chan-sccp-b.sourceforge.net/
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.16 (GNU/Linux
figured out how to create the stefan@.. solution.
To reach this context, people have to call IAX/gu...@my.asterisk.pbx
How do I create a context in which all calls from nonregistered clients
are handled?
- -S
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
On 06/09/2011 08:50 PM, Jamie A. Stapleton wrote:
Guest calls go to the context specified in [general] of sip.conf.
Thx. Is this valid for IAX2 also?
- -S
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 05/18/2011 06:23 PM, Jason Parker wrote:
To clarify, does that mean that you were able to successfully use
facebook chat with sasl?
This is correct.
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
I finally figured it out.
For facebook chat to work you have to use
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 04/17/2011 02:28 AM, Stefan Gofferje wrote:
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com
But the hint doesn't show any difference. It always shows online
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
-S
- --
(o_ Stefan Gofferje| SCLT
with a
gigabit-link to the switch.
That should get the line quality pretty much to the bottom.
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
signature.asc
Description
On Sunday 17 April 2011, Stefan Gofferje wrote:
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP
On Sunday 17 April 2011, Stefan Gofferje wrote:
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com
But the hint doesn't show any difference. It always shows online on the
phone and core show hints always shows
?
Asterisk SVN-branch-1.6.2-r313579
- -Stefan
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.16 (GNU/Linux
...?
- -S
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.16 (GNU/Linux
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
- -S
- --
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
Sasa schrieb:
I need other files other than those obtained with
cmterm-7911_7906-sip.8-0-4sr1.cop ??
cmterm is the callmanager software. You need to get the non-callmanager
SIP-software. Contact your local Cisco representative to buy a license
for that.
Terve,
Stefan
--
Last words of a
Asterisk Development Team schrieb:
[Release info]
Did anyone notice bug #0013531 (http://bugs.digium.com/view.php?id=13531)?
It seems that the hold logic / MOH logic in chan_sip is somehow broken
in 1.6.0...
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
Michael Graves schrieb:
Earlier today I glanced at Junction Networks blog and was surprised to
find a post indicating that Cisco was dropping SIP support in their
79xx series phones. Here's t
link:
http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo
Hi,
[EMAIL PROTECTED] schrieb:
I have done what you told me to do, but nothing changed. Always the same
problem.
If I understand your dialplan right, your * is still calling itself via
SIP, right?
This is what is called a loop. You should review your dialplan and
replace all dial(SIP/[EMAIL
Hi,
[EMAIL PROTECTED] schrieb:
In fact, after entering in Asterisk for the first time, my call is
redirected to an other component of my system. This other equiment
redirect the same call to Asterisk a second time.
Hm, I suppose, your equipment is using reinvites for that redirection.
The
Hi,
I have the following symptoms:
Call X-lite / Nokia E51
X-lite press hold: Nokia DOES hear MOH
Nokia press hold: X-lite does NOT hear MOH
Call X-lite / SCCP phone
MOH works as supposed
Call SCCP phone / Nokia E51
SCCP press hold: Nokia DOES hear MOH
Nokia press hold: X-lite does NOT hear
Hi,
[EMAIL PROTECTED] schrieb:
I have a SIP request which comes from an Asterisk and which has to
re-enter in the same Asterisk (during the same session), but during the
second passage in Asterisk, it send me a 482 Loop Detected. So is it a
bug or Asterisk control the session and considere
[EMAIL PROTECTED] schrieb:
Thanks for help, but I don't understand what you say. How is it
possible to handle the error in the dialplan if my request return a 482
after entering Asterisk, but before accessing the dialplan ?
Ok :). I meant, you should handle the whole thing in the dialplan
[CC] chan_agent.c - chan_agent.o
chan_agent.c: In function ‘unload_module’:
chan_agent.c:2496: error: void value not ignored as it ought to be
make[1]: *** [chan_agent.o] Error 1
make: *** [channels] Error 2
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
Hi,
when I still had ISDN, I was using Hangup(causecode) to send e.g. Wrong
number to unwelcome callers.
Meanwhile, I am only using SIP providers (no PSTN lines any more) and I
would like to do similar, i.e. send specific SIP headers. Besides wrong
number, I would especially like to send 302 temp
Remco Barendse schrieb:
Suprising that this feature isn't used much, i would suspect that many
asterisk installations (including mine) have very simple (short) extension
numbers which makes brute forcing them rather easy.
Extension numbers and SIP account basically have nothing to do with
Barton Fisher schrieb:
It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP
aware. Apparently, this current
firmware/programming is not, one way audio problems.
Is there a version that support VoIP directly for this router?
Do you have firewall feature set? Then you
Kristian Kielhofner schrieb:
IMNSHO, the less SIP aware the better...
I have to disable SIP inspection on every IOS/PIX device I come
across. Fix the one-way audio problems on your proxy, registrar, etc
(in the case, Asterisk).
Most SIP ALGs are broken.
Interesting. I have my
No worries - I forgot a smiley. I didn't mean to appear annoyed or
otherwise negative.
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
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http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
-- Bandwidth and
Alex Balashov schrieb:
The short answer is SIP.
Maybe not behind a firewall which you don't have control over. IAX is a
single-port-protocol and as such much less problematic with firewalls
and NAT.
Read the second link in my previous mail.
Terve,
Stefan
--
Last words of a stormchaser:
Where
Julien Claassen schrieb:
IAX I can basically understand, although I wasn't aware in the slightest,
that other standard softphones supported it.
They don't. Well - it depends, what you see as standard. There are very
good multi-platform combined SIP/IAX clients like Zoiper. But Zoiper is
not
Gordon Henderson schrieb:
So comments, ponderings or anecdotes, etc. ... ?
Bristuff worked perfectly fine for me for about 5 years.
HOWEVER, you should keep in mind, that bristuff are very extensive
patches against the zaptel dirvers and also against the core. So
regarding updates you are
Hi,
did anybody get SIP TLS working with E51?
If I enable security in the phone's SIP config, the E51 attempts a
REGISTER via 5060 UDP with method TLS, digest. My asterisk (latest
SVN) just answers 401 UNAUTHORIZED.
Is there some comprehensive howto for configuring SIP TLS?
Terve,
Stefan
--
Hi,
I would like to arrange that when an IAX client logs in / registers with
my * AND there are unread voicemails, this IAX client will be
automatically called and connected to the respective voicemail box.
One possibility is to have a cronjob that creates a callfile - let's say
- every five
RoLaNd RoLaNd schrieb:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever
Hi,
I was switching from zaptel to dahdi and got latest SVN from everything.
Compiling works fine.
kernel module dahdi_dummy is loaded.
/dev/dahdi/pseudo exists
Trying to go into a meetme does not work:
[Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1]
MeetMe(SCCP/6000-0001, 444|dcIM)
Kevin P. Fleming schrieb:
Fixed in revision 137188; this module apparently did not get any DAHDI
conversion work at all, but I don't know how it got missed. Thanks for
the testing!
Confirmed. Works fine now under all (extensively) tested conditions.
Terve,
Stefan
--
Last words of a
Russell Bryant schrieb:
You'd have to provide a packet capture to see exactly what is happening.
It sounds like on the call leg between your client and Asterisk, it
isn't offering encryption as a capability, so it doesn't get used.
However, when your friend calls you, and Asterisk makes
This is how it sounds:
http://stefan.gofferje.net/chan_mobile_distorted.wav
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
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AstriCon 2008
Russell Bryant schrieb:
Interesting. Here are a couple more sanity checks you can do. First,
double check to ensure that your entry in iax.conf has encryption=yes
set. Also, when you make the call into Asterisk, set the verbose
setting up a bit. You should see output from chan_iax2
Hi,
addons 1.6 don't compile here. Any ideas?
Terve,
Stefan
[EMAIL PROTECTED]:/usr/src/asterisk-addons-1.6.0-beta4 make
CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent makeopts
make[1]: Entering directory
`/usr/src/asterisk-addons-1.6.0-beta4/menuselect'
Hi,
Russell Bryant schrieb:
It looks like you're trying to compiled Asterisk-addons 1.6 against
Asterisk 1.4. You will need to install Asterisk 1.6 before you can
compile and install Asterisk-addons 1.6.
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it
Stefan Gofferje schrieb:
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it readily compiled in the neighbour dir?
Confirmed - works. Thank you!
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar
Russell Bryant schrieb:
Stefan Gofferje wrote:
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it readily compiled in the neighbour dir?
That is correct, at least for the easy case.
Alternatively, you can specify the Asterisk location as an argument
Hi,
I started testing chan_mobile. Target is having some old phone with a
duosim (second card with same number) put to silent somewhere in the
rack with the *. That phone should mainly take incoming calls and after
45secs put them to the mailbox AND permit me to talk via my nice Cisco
desktop
Hi,
I have configured all IAX clients with encryption. I use Zoiper as a
softphone. When I make a call in the LAN from desktop-PC to *, the call
is - according to wireshark not encrypted. Wireshark identifies the
packets as normal G.711 mu-law packets. However, * reports the client as
encrypted:
That does not make too much sense to me... Configuration should be ok...
[Aug 8 23:30:13] SSL certificate ok
[Aug 8 23:30:13] == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd:
FILE * open failed!
Hi,
is there meanwhile the possibility for some actions besides dialling in *?
Namely, I would like that if a remote IAX or SIP user logs in AND there
are new messages, they automatically get a call and be connected to the
voicemail. The only method I know by now is make a context in the
Olivier schrieb:
Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to
a SIP call.
Is it certain ?
Yes, just tested it myself. Phone answers with Busy here if in a GSM call.
From my understanding, Symbian applications MUST leave this decision
type to an external
Olivier schrieb:
As a dual GSM/WiFi mode phone might be already engaged in a GSM
conversion while a SIP call occurs, I think Symbian application dev
rules would impose any application to centralize microphone and speaker
allocation to a Symbian provided resource manager.
So I think
Olivier schrieb:
Which company publishes this Symbian application implementing a local
answering machine on the phone, for instance ?
There are several. For instance, rock your mobile comes to my mind.
http://www.rock-your-mobile.com/
http://www.rock-your-mobile.com/answering-machine.php
-S
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
--Stefan
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix,
Gordon Henderson schrieb:
On Mon, 4 Aug 2008, Patrick wrote:
Hi Stefan,
Stefan Gofferje wrote:
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
I looked into this for my N95 but if it's possible then it isn't
Gordon Henderson schrieb:
Seriously - Push to talk - Half duplex Communications. How ancient is
that! It's really reminiscent of ancient US style truckers - Smokey and
the Bandit and all that. That's just so last century. Lets put all that
behind us and get with the 21st century! We all
Hi,
when I put a call on hold from my Nokia E51 (SIP client), the other side
does NOT hear music on hold although sip debug / wireshark shows that
the E51 tells the asterisk that it now holds the call. Canreinvite is
set to no.
Also, when parking a call (features.conf), the parked caller does not
timeout shows, that the PSTN isn't satisfied with just what Progress()
sends.
Does anyone got this Early-B3 working with BriStuff and HFC-S?
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point
to
the skinny phone.
http://chan-sccp.org/
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
--Bandwidth and Colocation sponsored by Easynews.com
Hi folks,
I understand that IAX2 supports public key authentication. Is the
transmission also encrypted or is it possible to encrypt an IAX2 trunk
between 2 *s?
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler
-users
Webforum:http://forum.chan-sccp.org/
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
--Bandwidth
of people working at various howtos at the moment.
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
--Bandwidth and Colocation
Wayne schrieb:
Stefan Gofferje wrote:
Hi folks,
whoever owns a Cisco phone and is unhappy about slow firmware,
incomplete XML support etc... should really have a look at Sergio
Chersovani's rewrite of chan-sccp!
Ok - I'll give it a go :) - Just one problem... My phones have been
documented sample config and - as I wrote before - there
are lots of info in the chan-sccp-users mailing list archive.
I myself started with a SIPped 7960 but meanwhile, I run 3 7960s and 1
7905 without any problems.
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd
someone tell me an outlet
where I could purchase a Smartnet contract to download this firmware? I
have been unable to find a retailer online that can help. Thanks in
advance for any help.
http://tools.cisco.com/WWChannels/LOCATR/jsp/partner_locator.jsp
Regards,
Stefan
--
(o_ Stefan Gofferje
browse the mailinglist archives at
http://lists.berlios.de/pipermail/chan-sccp-users/
Probably a stupid question, but does this thing have a backlight?
7960? Nope... 7970 has.
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_
not only supports call
forwarding but a lot more and only with the Skinny image, you can use
all the features, the phone have.
You can read a bit more at http://chan-sccp.org/
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_
-sccp.org/ ...
Slan go foil,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
--Bandwidth and Colocation sponsored by Easynews.com
several times. Same is with echo test (call taken from PSTN)
Get a CAPI module for your Teles and try chan_capi-cm from
http://sourceforge.net/projects/chan-capi/
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch
) or chan-sccp.org
(unofficial site). There is a related mailinglist at berlios.de where
Sergio does a hell of a lot of support (unless he is one vacation like at
the moment :-) ) and gladly accepts bug reports :-).
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux
On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote:
Hi,
I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp.
Jep, it is... If you had problems with this, your chance for a solution is
higher at the chan-sccp-users list... :-)
Regards,
Stefan
from all the hot bug
fixing :-)
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
Asterisk-Users mailing list
Asterisk-Users
hangupcause you want
back to the network e.g. 021 call rejected :-) ...
chan_capi-cm - http://sourceforge.net/projects/chan-capi/
cause codes - http://www.telos-systems.com/?/techtalk/cause.htm
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler
= client_int_unrestricted
callwaiting = 1
incominglimit = 2
mailbox = 1000
vmnum = 8500
cid_name = Bedroom
cid_num = 6004
line = 6004
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point
Cisco with
a Cisco T1-card...
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
___
Asterisk-Users mailing list
Asterisk-Users
testing here. Until now, I just found smaller
inconsistencies, no real nasty bugs.
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch - the original point and click interface
Hi folks,
when I start asterisk directly, I get a colored CLI. When connect to a
already running asterisk with asterisk -R, it's never colored, despite
I'm running both from the same console (tty). Is there a way to force
asterisk -R into color mode?
Regards,
Stefan
--
(o_ Stefan
[EMAIL PROTECTED] schrieb:
Asterisk -gc
I don't see a -R in that...
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch - the original point and click interface
, on incoming calls, the Siemens shows the CallerIDName as set by
Asterisk in the display. zaphfc also supports SendText...
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch - the original
any hints on that?
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch - the original point and click interface
___
Asterisk-Users mailing
it happen with plain 1.0.7 as well
Don't know. My * is kinda productive, so I am bound to BRIstuff for use of
zapHFC for internal phones.
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch
On 11:28:09 June 27, 2005 Armin Schindler [EMAIL PROTECTED] wrote:
Yes, maybe you would like to implement some of those feature-requests
? ;-)
I would love to if I were a bright programmer :-).
--Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User
Armin Schindler schrieb:
On Sat, 25 Jun 2005, Stefan Gofferje wrote:
Armin Schindler schrieb:
I have added busy()/congestion() support to CVS HEAD now, can you please
test if it works for you?
Works perfectly well! Also CallingPres(32) does work! The only thing I wonder
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