4 sep 2009 kl. 08.05 skrev Gordon Henderson:
On Thu, 3 Sep 2009, Asterisk Security Team wrote:
+
+
| Discussion | A lot of time was spent trying to come up with a
way to |
|| resolve this issue
4 sep 2009 kl. 13.40 skrev Marius Ciorecan:
Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
which I connected an external PSTN line. I use it as carrier for VoIP
calls. I can make successfully calls, but there's one problem, I
receive
200 OK with SDP with delay
4 sep 2009 kl. 14.02 skrev Amatisoft SRL:
Description
* Amatix Office is a complete communication platform for small and
medium-sized businesses. It provides email, calendar, contacts,
conventional telephony like ISDN, new generation VoIP telephony like
SIP and more other features
4 sep 2009 kl. 16.05 skrev David Budny:
You can do this through your dial plan by using the “failed”
extension within your context.
Add this to your dial plan:
Exten = failed,1,NoCDR()
When using a call file and the call fails, the call will jump to the
failed extension within your
4 sep 2009 kl. 21.00 skrev Jerry Geis:
Hi all
I have asterisk 1.4.12 that was working on CCM 4.0
they updated to 6.1.3 and it no longer works.
I tried updating to 1.4.26.2 but still not working.
I get SIP error 503 service unavailable.
From Cisco or from Asterisk?
/O
acceptable for
different codecs.
From sip.conf.sample:
;allow=ilbc ; see doc/rtp-packetization for
framing options
/O
---
* Olle E Johansson - o...@edvina.net
* Open Unified communication - Asterisk, Kamailio, Sip-router projects
3 sep 2009 kl. 00.27 skrev John A. Sullivan III:
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
i have posted this before but was unable to resolve it. i have some
new info so i figured i would try again. the trace from bandwidth.com
are below. they are telling me that the ip that is
1 sep 2009 kl. 08.17 skrev James Mutuku:
Hello,
From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer,
it says that there For Asterisk 1.2 there was no jitterbuffer in the
RTP-based channels (i.e. chan_sip).
I am using 1.2 and Ind there is no reason to upgrade. Are there
1 sep 2009 kl. 05.18 skrev John A. Sullivan III:
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote:
Hello, all. In our multi-tenant environment, we would like to be
able
to use the reinvite media redirection within Asterisk for calls
within a
tenant but not between
Friends,
I would like to congratulate kamailio.org - a project we're
cooperating a lot with. They have just been awarded the BOSSIE award
by InfoWorld. Kamailio is the OpenSER SIP proxy project with a new
name, a product widely used in Asterisk installations. And of course,
the motivation
2 sep 2009 kl. 02.44 skrev James Lamanna:
Hi,
I'm trying to configure 2 parallel sip trunks between 2 boxes.
However I seem to have the problem that when making a call from Box 2
to Box 1, it sometimes
says authentication failed because it is using the username of the
other trunk.
27 aug 2009 kl. 11.24 skrev Klaus Darilion:
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP statistics out of Asterisk
.
/Olle
---
* Olle E Johansson - o...@edvina.net
* Open Unified Communication - SIP XMPP projects
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25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit is 11000 channels or 5500 calls, isn't it ?
Yes, this is 11.000 channels.
Bonding is good advice, provided we
25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit
25 aug 2009 kl. 19.42 skrev Steve Totaro:
On Tue, Aug 25, 2009 at 12:28 PM, Olle E. Johansson o...@edvina.net
wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further
21 apr 2009 kl. 11.46 skrev Khaled W. Chehab:
Dears,
When my GW send a call to asterisk v 1.4.24 ,
Asterisk send Status: 420 bad extension (unsupported)
Why? Any modifications should be done one sip.conf
regards
Your gateway is propably requiring a SIP extension Asterisk does not
7 apr 2009 kl. 18.26 skrev Florian Hackenberger:
On Tuesday 07 April 2009, Olle E. Johansson wrote:
I don't see any problems there. YOu still have devices with states,
as you would have with authentication. Of course, it still depends on
your configuration. But authentication should
6 apr 2009 kl. 09.58 skrev Florian Hackenberger:
Hi Philipp!
On Sunday 05 April 2009, Philipp von Klitzing wrote:
Take a look at these two links:
Thanks for the links! So one option is to implement domain based
authentication, which would be quite a bit of work. Another option
which is
it with whatever was sent in the To: header
(which should be the original destination) before hitting the dialplan.
That code still exists in a branch somewhere and in Pineapple.
This code would also solve the issue with registering multiple
accounts with one provider.
/O
---
* Olle E. Johansson - o
7 apr 2009 kl. 11.49 skrev Florian Hackenberger:
On Tuesday 07 April 2009, Olle E. Johansson wrote:
Well, you can have OpenSER doing the authentication and turn it
off in Asterisk, but still match a device.
Ok, but what about sip device state? Will that work? Will asterisk
report the device
7 apr 2009 kl. 12.08 skrev Steve Davies:
2009/4/7 Olle E. Johansson o...@edvina.net:
[snip]
The REGISTER request in the RFC was really written for a device.
The way providers use it for trunks with multiple DIDs is outside
of the
RFC and is discussed in relation to the SIPconnect
4 apr 2009 kl. 19.31 skrev Benny Amorsen:
Martin asteriskl...@callthem.info writes:
yes, that's Asterisk's problem but it seems OP is talking here about
something else that produces that particular
message check_auth: username mismatch, have 7705, digest has
7736
I really believe it's
Plack
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* Olle E Johansson - o...@edvina.net
efforts in trying to help, it would be better
not shooting from the hip ;-)
/O
---
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* Asterisk Training http://edvina.net/training/
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the other end sends it from.
/O
---
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* Asterisk/OpenSER/Kamailio Training http://edvina.net/training/
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2 apr 2009 kl. 20.42 skrev Kevin P. Fleming:
Danny Nicholas wrote:
You should not have a G729 command on the CLI. Codecs are
addressed in
sip.conf, dahdi.conf, etc. restarting Asterisk might do the
trick. You
only need to reboot for a driver level change.
This is incorrect.
/mailman/listinfo/asterisk-users
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* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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3 apr 2009 kl. 15.14 skrev Kevin P. Fleming:
Olle E. Johansson wrote:
Is this also available as a manager command?
I would really appreciate being able to check license status over
manager.
It is not today, but I'll make a note to add it to the next builds,
which will probably happen
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
In a surprising move, Digium in partnership with Edvina today released
a new channel driver for Asterisk, chan_tweet. The driver connects
seamlessly to several microblogging platforms, including
.
This was actually caused by a bug in some version of Asterisk 1.4,
which I fixed. If you update it should not happen.
/O
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17 mar 2009 kl. 07.26 skrev zoach...@securax.org:
Vincent Li wrote:
Hello,
I just had a meeting about a pilot project going on in our
University, The
project manager has done some research in the past year and
concluded that
Asterisk can not scale well to large user base like 10,000
6 mar 2009 kl. 13.36 skrev Mikel Lindsaar:
Hi all,
Is there any way to make use of the SIP making progress messages?
I find that about the time the SIP peer says making progress is the
time the other end actually starts to ring, or is busy etc.
Before that time, I want to generate an in
17 dec 2007 kl. 10.45 skrev randulo:
On Dec 15, 2007 11:57 AM, Johansson Olle E [EMAIL PROTECTED] wrote:
I'm kind of interested in the slow uptake of Asterisk 1.4. Between
1.2
SNIP
- Not enough reasons to upgrade, since 1.2 really works well
- Just a bad karma for 1.4
Hi Olle,
It's
Tony,
Thanks for the feedback!
17 dec 2007 kl. 12.40 skrev Tony Mountifield:
- I have a lot of customisations to app_meetme, which I will need to
port
How about sharing them so we can maintain them in the open source base?
:-)
- Scare stories about IAX-related lockups in 1.4, due to the
17 dec 2007 kl. 14.11 skrev Tilghman Lesher:
On Monday 17 December 2007 04:17:32 Thomas Stein wrote:
The only problem i have is with sending and recieving Faxes. Right
now i'm
using spandsp an app_rtxfax. This works fine. But there seem to be no
spandsp and app_rtx packages in my gentoo.
17 dec 2007 kl. 15.26 skrev Tony Plack:
- Not enough reasons to upgrade, since 1.2 really works well - Just
a bad karma for 1.4
Funny, but my results have been different. I was running on 1.2.17
(and on to 22) for a year and had all sorts of lockups. For me,
when I switched to 1.4.5
17 dec 2007 kl. 15.42 skrev Tony Plack:
All I can say is with 1.6, if a change is made that causes
something that worked in 1.4 not to work in 1.6, please think
twice, three times or four times before making the change, or
making the change in such a way that it won't break dialplan
stuff
However 1.4 since release have had some serious changes that blocked
our
planned upgrades - for example some memory corruption that raised
between 1.4.10 and 1.4.12 that was very hard to track down. This shows
that having 1.4 in bugfix-only state is not actually working that
good -
we
Agreed.
Given that our group has many 1.2 versions working well on CentOS 3.x
boxes, and that 1.4 requires either 4 or 5, your option of starting
all
over is about all that will work.
I would like to know a bit more on why Asteirsk 1.4 means that you have
to upgrade Centos? (obviously not
17 dec 2007 kl. 18.57 skrev Olivier:
Hi,
To summurize, it seems that one thing preventing people from
upgrading is the lack of an upgrading tool : somehow, it should be
possible and easy to :
- install 2 different versions of Asterisk on the same hardware,
- interactively translate
17 dec 2007 kl. 19.33 skrev Ira:
At 02:55 AM 12/17/2007, you wrote:
I wonder if there are any major obstacles for upgrading.
Because of your message I tried upgrading to 1.4 again Saturday. That
was the third or fourth time I've tried and the first time it's
lasted more than a few hours
17 dec 2007 kl. 18.49 skrev Roger Schreiter:
Hi,
some months ago, I had the problem with an asterisk-1.4.x-
Version, that some calls (but not all) were interrupted
64 seconds after connect (a call limit of 86400 seconds
was installed using the S()-parameter).
It was just a test machine,
17 dec 2007 kl. 21.00 skrev shadowym:
I do wish Digium or whoever tests this stuff had a more reliable way
of
testing software releases rather than relying on feedback from the
community. Fonality, for example use what they call a hammer
which sounds
to me like a bunch of servers
We run everything on ubuntu server 6.06 LTS and also use freepbx as
the
interface with some minor customisations. It works very well and we
are
now shifting some others to 1.4 but the issue is if anything goes
wrong
its too costly to fix, as part of maintenance we keep them
remember that
confidentiality of the media
stream is only one small piece of the larger VoIP security puzzle.
Even if the media is
encrypted by ZRTP, signalling might reveal information that you
consider private.
/O
---
* Olle E Johansson - [EMAIL PROTECTED]
* Asterisk SIP masterclass, Stockholm
15 dec 2007 kl. 14.48 skrev Olivier:
Hi Olle
2007/12/15, Olle E Johansson [EMAIL PROTECTED]:
14 dec 2007 kl. 11.20 skrev Andres Gomez:
Hello List
I am very interested in developing a research project on security
protocol for VoIP, under the GPL.
For some time I have been
15 dec 2007 kl. 15.42 skrev Steve Totaro:
Johansson Olle E wrote:
Friends in the Asterisk community,
I'm kind of interested in the slow uptake of Asterisk 1.4. Between
1.2
and 1.4 there's been a lot of
important development. New code cleanups, optimization, new
functions.
I realize
All I can say is with 1.6, if a change is made that causes something
that worked in 1.4 not to work in 1.6, please think twice, three
times or four times before making the change, or making the change
in such a way that it won't break dialplan stuff from 1.4.
Our policy is to never
It seems that all the warnings about deprecated functions in 1.2 did
not give the desired effect - that users move from the 1.0 commands to
the new applications and functions in 1.2. That caused real problems
when going from 1.2 to 1.4, since the dialplans where still on 1.0
level, not 1.2
My biggest gripe is that everything loaded and seemed to work. A
day later we found this did not work and discovered a syntax
change. A day later something else did not work, an other syntax
change. Why isn't there some pre-processor to check the syntax of
the config files? Would
The Asterisk BOF session will be tomorrow at 4.30 PM, VON Stockholm
at the Stockholm Fairgrounds in Älvsjö.
http://tinyurl.com/3degv5
From time to time you will find me in the Voop stand in the Digium/
Asterisk Pavillion in
the exhibition.
See you!
/Olle
-
Sponsor Codename Pineapple
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
25 maj 2007 kl. 06.40 skrev JK:
Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to
set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not
working.
In our scenario the SP is sending call to our ser server and ser
is forwarding the
17 maj 2007 kl. 02.57 skrev Robert Lister:
On Wed, May 16, 2007 at 05:46:21PM -0500, Greg Oliver wrote:
If I push the response code back to the handset (Cisco 7960) then
it is even
more unhelpful as it uses the same error message for all SIP
error type
response codes: Reorder but does not
13 apr 2007 kl. 16.45 skrev Brian Jones:
I've encountered a similar problem with Cisco equipment. The Cisco
proxy was not replying to Asterisk with an ACK after * sent an OK.
Since version 1.2.14, * was changed so that not receiving an ACK to
an OK is considered a FATAL error.
The
Friends,
I have gotten a few questions lately on the status on the Codename
Pineapple project, the project
that hopefully will produce a more stable and SIP compliant SIP stack
for Asterisk.
Due to lack of funding, it's postponed until further notice.
I have a few sponsors, but not enough
8 maj 2007 kl. 16.57 skrev SIP:
Joshua Colp wrote:
Alex Lake wrote:
I understand that it is customary for SIP User Agents to send
OPTIONS packets every now and then to check that a peer is still
alive and well. Indeed I understand that Asterisk itself sends
them if qualify is set to yes
9 maj 2007 kl. 14.46 skrev Steve Blair:
[May 9 08:42:43] WARNING[20530]: pbx.c:1783 pbx_extension_helper:
No application 'SIPGetHeader' for extension (default, 700, 4)
Well, Asterisk is trying to say something to you. Did you listen?
SIPGetHeader is now a function called SIP_HEADER
We do
9 maj 2007 kl. 18.14 skrev Ken Williams:
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The
problem start, once a week or so the SIP phones couldn't
communicate with the server, though there was no error
10 maj 2007 kl. 18.29 skrev FUERMANN, JASON BRYCE:
I have also only tested but it did work well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Thursday, May 10, 2007 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial
8 maj 2007 kl. 15.40 skrev Joshua Colp:
Rohan Hathiwala wrote:
Hi,
I need asterisk to instruct the other side to send RTP to a
conference
server running on a different machine. The conference server does not
understand SIP so I cannot use the SIP REFER method.
I have another question.
We're actually getting two invites and schedules retransmit of both,
which is bad. One retransmit is stopped and the other one keeps
going, regardless of the ACKs that keep coming in. Needs to be fixed.
Believe I have fixed this in 1.4 svn, please test.
/O
- Patch
Index:
3 apr 2007 kl. 09.07 skrev Raj Jain:
Olle,
It depends on how strictly the UA adheres to the offer/answer
model. The issue would be that a RE-INVITE from Asterisk will have
the version number incremented by more than one, which will break
the following rule.
Quoting from RFC 3264
3 apr 2007 kl. 10.04 skrev kjcsb:
The call that gets dropped had a retransmission of INVITE from UAC
to UAS (and therefore retransmission of 200 OK from UAS to UAC).
There is nothing wrong with the re-transmission as such, but I
noticed a potential bug in Asterisk in the way it responds to an
5 apr 2007 kl. 13.04 skrev Raj Jain:
Regarding project Pineapple, I'm curious why rewrite (or refactor)
the SIP stack instead of using an open-source one. Did your
research show that there is nothing viable out there that'll fit
well w/in Asterisk? OpenPBX community is talking about
-Agent: Blah
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
Your server is sending a NOTIFY that the ITSP's server doesn't like.
Propably a mailbox notification.
Not a critical error, just a configuration issue.
/Olle
---
* Olle E. Johansson - [EMAIL
it...
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP master class, Stockholm may 2007 - register now!
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2 apr 2007 kl. 13.50 skrev sravana:
Anybody done LDAP authentication in Asterisk? can you explain how?
Thanks in advance
There's some code available in the issue tracker. Please check in
bugs.digium.com
for res_auth
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training
2 apr 2007 kl. 19.32 skrev Raj Jain:
I found a subtle difference between the two traces you sent (the
call that works and the call that gets dropped). This may or may
not be what's causing the problem.
The call that gets dropped had a retransmission of INVITE from UAC
to UAS (and
registrations from asterisk and give it to SER
directly or other registrar server.
Using the realtime subsystem, you can share registration data between
Asterisk servers. In combination with
Dundi and the regexten= system, it's even more dynamic.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED
and needs to be enabled.
You are assuming that SIP works like zaptel in the dialplan, but it
does not. You propably need to re-configure your phones.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP master class, Stockholm may 2007
Runnstam at PulseVoip in Bergen,
Norway.
* Asterisk SPITwall(R) - filtering away tomorrows VoIP spam today!
--
The SPITwall(R) technology is developed by Olle E. Johansson, a
member of the Asterisk developer
27 mar 2007 kl. 10.48 skrev Tim Panton:
On 26 Mar 2007, at 22:32, Michael Graves wrote:
Hi All,
I've been reading about Phil Zimmermann's ZRTP encryption scheme for
SIP clients. This seems attactive but I don't use soft phones. I'm
guessing that we'd need ZRTP support in Asterisk in
order
17 mar 2007 kl. 12.21 skrev Rizwan Hisham:
Hi,
I have declared my sip users call-limit=2 and type=friend. When any
user recieves a waiting call while already in a conversation, the
peer call counter is set to 2.The problem is that, the counter is
not reset to zero after hangup and becoz
12 mar 2007 kl. 23.33 skrev Nikhil Jogia:
Bruce Reeves wrote:
Does SIPAddHeader(Alert-Info:) not do it?
No, but from another thread, setting the _SIPADDHEADER variable works.
You misunderstand. The prefered way is to use SIPAddHeader(Alert-
Info: slakfj aslkfjaklsdf)
But in the
13 mar 2007 kl. 09.53 skrev Noc Phibee:
Hi
i have a big change or bproblems to update a asterisk 1.2.12 server
to asterisk 1.4.1 ?
There won't be any problems if you take some time to read the
available documentation
to see what changes you need to do in your configuration.
Make sure
Friends,
This week I'll be in Lissabon speeking at a Voip Conference on
Wednesday. I'm not aware if there's
an Asterisk Users group in Lissabon, but if there is maybe there
would be a chance to meet.
Next week, I'll be at Cebit, in the Digium stand. If you want to meet
me, I'll be in the
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12 mar 2007 kl. 19.16 skrev Octavio Ruiz (Ta^3):
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this
via AMI:
Action: Originate
Channel: Sip/1234
Application: AgentLogin
Data: 1234
Variable: _ALERT_INFO=info=alert-autoanswer
Callerid: AutoLogin[1234]
In order to send an
variable value (set in
configuration for account)
-
Please test Mini-Voicemail for Asterisk 1.2!
http://www.voip-forum.com for more information.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass, Stockholm, May 7-11
of that cocky
statement that Minivm
was using the same realtime handle as voicemail(). Not god. It's now
minivm.
Thanks for pointing me in that direction.
/O
Julian
Olle E Johansson wrote:
Just to tease users to test Mini-Voicemail:
*CLI show function MINIVMACCOUNT
-= Info about function
that each channel/media driver has different ways to handle
this.
Zapata has native bridging too, but different rules apply depending
on the
technology used.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass
8 mar 2007 kl. 14.36 skrev René Enskat:
hello all,
My problem if i have my extensions and sipusers in a realtime
database it is not possible to use BLF or hinting.
i see only idle or unavailable status but if the phone is ringing
or in use i can't see it.
Is there a fix or any workaround?
8 mar 2007 kl. 21.05 skrev Daryl Jurbala:
OK...that makes much more sense. So here's my follow-up question:
what's the easiest way to check if I'm native bridging a call. I'm
trying to offload as much RTP traffic as possible, and want to have
a way to check quickly (there are well over
7126
host=dynamic
nat=yes
canreinvite=no
allow=all
You have set canreinvite to no, thus disabling native briding.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass - Stockholm, Sweden, May 7-11
Off topic:
I usually joke with students about response codes to a SIP bye request:
What happens if you send a BYE and the other side responds 603
declined ?
- I don't want to hangup, I want to continue talking
Mother-in-laws would love that...
/O
8 mar 2007 kl. 11.20 skrev Andreas Anderson:
Hi,
on ISDN there are the numbering plans that indicate if it's an
national or an internation number. Is there something similar on
SIP? How should i set a callerid to an internation number? complete
e164, with, without an intl prefix (ie +,
4 mar 2007 kl. 21.45 skrev André Santos:
Hi,
I am implementing de Real Time architecture, I would like to know
if, their is any problem in putting the section [general] of the
sip.conf file in the table of sippeers.
You can't mix the general section with sippeers for realtime.
7 mar 2007 kl. 15.38 skrev nik600:
what is the support in asterisk for ssl voip protocols?
I am looking for a solutions to grant the possibility to some users to
use an asterisk server as a proxy voice, for talking each them in a
safe and secure mode on internet.
Is it possible?
No.
/O
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8 mar 2007 kl. 04.16 skrev Bill Gibbs:
Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16
“sip show channels”
Always tends to show 100+ lines such as
192.168.1.241(None) 2e2872da-1d 00101/21507 unkn
No Rx: REGISTER
Never seem to go away
198 total peers on this server
8 mar 2007 kl. 05.33 skrev voiplist:
Um, is it possible to patch 1.2.4? We have some pretty busy
production systems and are not exactly excited about having to upgrade
from this version.
Is there no other way to protect our systems from this hole?
You can apply the patch that was applied
Friends in the Asterisk community,
One thing I avoided working with for a long time is the Asterisk
voicemail code. One module in
Asterisk I've constantly been naming as one of the worst parts is
voicemail. One part of
Asterisk that I've been kind of avoiding during my trainings is
Everything can be done with a certain amount of coding... :-) No,
it's not possible
in Asterisk today.
Check the configuration templates to make life easier when
configuring voicemail.
It's documented in doc/configuration.txt in your asterisk source code
directory, or
here:
23 feb 2007 kl. 09.52 skrev Michiel van Baak:
Hey,
We have asterisk 1.2.4 (old I know) with a couple of snom
phones, a couple of grandstream phones and around 65 philips
dect stations.
Now the problem:
All calls do peer to peer RTP except the calls from dect
station to dect station.
snom to
23 feb 2007 kl. 12.42 skrev Steve Davies:
Hi,
In older versions of asterisk I used to be able to use
incominglimit=1 to effectively disable call waiting on a specific
SIP channel (Where broken phones do not allow this on the handset
itself)
In 1.2.x this became call-limit=1, but this
23 feb 2007 kl. 14.06 skrev ast guy:
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
Yes, Asterisk is a multiprotocol Open Source PBX. Those
functions
23 feb 2007 kl. 19.55 skrev Philipp Kempgen:
Olle E Johansson wrote:
22 feb 2007 kl. 23.40 skrev Philipp Kempgen:
Olle E Johansson wrote:
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
I thought it might be useful to be able to ask Asterisk for the
current SIP CSeq through the Manager API
24 feb 2007 kl. 03.15 skrev Yuan LIU:
How do I receive text sent from SendText() application? Asterisk
lists text capability, so SendText() is successful. But I don't
see an application to actually use it.
EyeBeam and several SIP phones does receive those messages.
We need to make sure
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