On 27.03.2014 10:39, jg wrote:
Wouldn't it make more sense to handle this by just monitoring the calls
and doing everything else with normal data processing?
Basically yes, but the whole idea is a workaround to fix issues in
legacy systems.
klaus
--
In my experience iaxmodem + Hylafax is very stable and work in my setups
fine.
But in these setups I either use ISDN uplinks or SIP trunks with low RTT
and jitter (highspeed links to the service provider, no WAN links).
Thus, T38 is in my setups not necessary.
regards
Klaus
On 24.03.2014
On 26.03.2014 11:30, jg wrote:
What do you mean with voice recorders? Voice mail, if nobody answers,
or do want to monitor calls?
I mean voice recording (just like a voice box does), but the recorder is
not Asterisk, but a dedicated VoIP recorder.
regards
Klaus
--
Hi!
I have strange requirement: a incoming call should be duplicated to two
outgoing calls (to two voice recorders). On the incoming channel we only
receive RTP, on the two outgoing channel we only send RTP.
I thought of:
incoming call
- originate: make outgoing call to recorder 1 and
Hi!
I want to forward a call to another destination if the outgoing call leg
has an rtptimeout. But as far as I see there is no way to find out if
the hangup was due to a rtp timeout or any other reason. I thought that
HANGUPCAUSE or DIALSTATUS would be set, but they aren't.
Are there any
Hi!
If you haven't noticed yet, SER (the mother of the SIP proxy projects
Openser, Kamailio, sip-router, opensips, ) is celebrating their 10th
year. There will be a main event happening in Berlin
(http://sip-router.org/10-years-ser/).
For those who can not travel to this event, there will be
Hi!
FYI: There is now an official IANA registration to map phone numbers to
IAX URIs.
http://tools.ietf.org/html/rfc6315
regards
Klaus
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Am 08.03.2011 11:05, schrieb Rizwan Hisham:
Hi all,
I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3
with cdr_odbc.
This is the best supposed solution i have come up with. But, I am here to
ask you people for your ideas and thoughts on my solution. I am still in
Hi!
Since some time the download of the newest Asterisk does not contains
the version number anymore, but is just called asterisk-1.4-current.tar.gz
This gives me a tarball where I do not know the version without looking
into the tarball.
Thus, IMO it would be very useful to switch back to
Am 10.08.2010 07:44, schrieb kamrun nahar bina:
Dear all,
What is the difference between SIPp and SER(Sip Express Router)? Which
one is better load performance testing?
Is there any one who knows about this? Could you please give me details
informtaion?
SIPp is a SIP test tool
Maybe we can easily extend the tool to crash Asterisk too (using some
exploits non-up2date Asterisk installations) ;-)
Am 24.06.2010 13:36, schrieb Randy R:
Hi,
Got some great news a few days ago from Sandro Gauci (@SandroGauci)
and we'll be talking about this with him this Friday at 1PM.
Hi Mike!
You are using wrong wording - with TAPI driver usually the TAPI service
provider is meant, e.g. see http://www.ipcom.at/en/telephony/siptapi/tapi/
So, the TSP offers lines to the TAPI subsystem. These lines can be used
by TAPI applications. Typical TAPI applications are dialer.exe
Am 01.06.2010 14:40, schrieb Mike:
figure out something. Meanwhile, if someone wants who has experience with
TAPI services wants to offer me his (paid) services I would be glad to
consider.
Hi Mike!
It depends on what you are looking for. I am the author of SIPTAPI and
have quite some TAPI
Hi Olivier!
There is a php AGI script available at
http://www.enum.at/index.php?id=522 that
1. performs ENUM lookup for CLI and looks for vcard service
2. fetch vcard from URI
3. fetch name from vcard and set the callerid-name.
maybe this helps you to start with vcard parsing
regards
klaus
Am
Am 17.05.2010 10:46, schrieb Zhang Shukun:
Hello,
you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)
My question is how could i identify whether the caller or callee
hangup the phone first?
AFAIK
If you can call yourself via the provider just setup a dialplan which
spirals the call,e.g. from softphone call via provider one of your
numbers. Then incoming call route to your next DID, and so on, and after
some spiraling just connect the call to the Milliwatt() application.
Milliwatt is
Regarding functions and applications options, the only authoritative
source is the console:
core show application ...
core show function ...
regards
Klaus
Am 07.05.2010 18:37, schrieb Tim Densmore:
Hi Folks,
Is there a generally accepted Asterisk bible for current versions? I
poked around
This code is really ugly und hard to verify.
Please file a bug report at https://issues.asterisk.org/
thanks
klaus
Am 06.05.2010 23:54, schrieb Richard Kenner:
I can confirm that the following fixes my problem:
--- chan_sip.c (revision 261450)
+++ chan_sip.c (working copy)
@@ -10357,12
Am 30.03.2010 20:56, schrieb Richard Kenner:
You need promiscredir set to yes on sip.conf
And then what do I do in the dialplan? I.e., what context is the
redirect number interpreted in? Google searches on this issue show
inconsistent and contradictory information.
I usually set the
The disconnect is RECEIVED by Asterisk. So there is a problem with the
other party.
You are sending FACILITY - maybe the other party does not like FACILITY
and hangs up.
IIRC there is a setting in zapata.conf to enable/disable FACILITY.
regards
klaus
Am 10.04.2010 21:46, schrieb bruce bruce:
Am 18.03.2010 05:11, schrieb Olivier:
2010/3/17 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Am 17.03.2010 10:40, schrieb Peter den Hartog:
Hello,
I was wondering if the following was possible:
When somebody sends a fax
Am 17.03.2010 19:31, schrieb Matt Watson:
On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion
klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote:
Attached is an untested (I did not had the time yet) port to
Asterisk 1.4.29.1 (DAHDI). Maybe the modules need some
Hi Jeff!
Looks like the term native bridging is a bit overloaded.
Some text from channel.h:
-# When the call is answered, Asterisk bridges the media streams
so the caller on the first channel can speak with the callee
on the second, outbound channel
-#
Am 17.03.2010 10:40, schrieb Peter den Hartog:
Hello,
I was wondering if the following was possible:
When somebody sends a fax to my direct number 0101234567105 (my
extension will be 105) is it possible that Asterisk, or an addon sees
this as a fax, and e-mail the fax to me?
So everybody
Am 17.03.2010 00:40, schrieb Steve Murphy:
How about:
blacklist(${exten});
macro blacklist(the_exten)
{
switch(the_exten)
{
pattern +4390[01]: Hangup();
default: return;
}
}
Yes, that would work. I didn't knew the pattern
Am 16.03.2010 20:59, schrieb Edwin Lam:
Steve Underwood wrote:
Crashes of this kind are not uncommon, but the causes are:
- Multiple versions of libtiff installed in different directories
checked that. got only single version.
- Multiple versions of spandsp installed in
Am 12.03.2010 19:05, schrieb Alexandru Oniciuc:
Hi Steve,
the remote device is an Hylafax Server that does ECM.
How is Hylafax connected to VoIP?
regards
Klaus
The sending fax
device, that's attached to the ATA, is a Philips fax machine with ECM
enabled. If I send with the same machine
Am 16.03.2010 01:42, schrieb Jeff Brower:
Vikram-
http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to
directly flow between endpoints and completely bypass Asterisk. I would
like to know if
Am 12.03.2010 18:01, schrieb Steve Underwood:
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote:
Hello,
I need a hand in choosing a small ATA, even with one FXS port, that
should do only fax with T38.
I’ve tried Grandstream (ht286 model) but the faxes go out without ECM,
even if the Fax
These commands are also available for 1.4. Looks like the DUNDI module
is not loaded. Watch debug logging during module load for errors.
Try ldd /usr/lib/astersik/modules/res_dundi.so and watch for
unresolved dependencies.
regards
klaus
Am 16.03.2010 03:01, schrieb John Haigh:
Are there
Am 12.03.2010 13:17, schrieb Steve Davies:
Hi,
I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap
:) I was wondering if someone could point me at 3 things that I appear
to have lost?
1) ZapEC(off) - Is there an equivalent dialplan command to request no
EC on a channel
Hi!
I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
dialplan:
application call to Gosub affects flow of control, and needs to
be re-written using AEL if, while, goto, etc. keywords instead
What is the suggested replacement for an explicit Gosub() call? I use it
Am 15.03.2010 13:48, schrieb Kevin P. Fleming:
Klaus Darilion wrote:
Hi!
I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
dialplan:
application call to Gosub affects flow of control, and needs to
be re-written using AEL if, while, goto, etc. keywords instead
Am 10.03.2010 17:33, schrieb Kevin P. Fleming:
Klaus Darilion wrote:
That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they
are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow
configuration of different ranges for UDPTL and RTP (although it
shouldn't be a problem
Am 02.03.2010 13:29, schrieb Magnus Benngård:
Hi!
Did a setup of 2 peers as Klaus suggested, it worked thx!
Has anyone thought about the possibility to add multiple ip/hosts to
host=?
I my case: host=130.244.190.42,130.244.190.46 or
host=sip-corporate1.tele2.se,sip-corporate2.tele2.se
On 10.03.2010 16:35, Michelle Dupuis wrote:
We are coordinating a connection to a SIP provider who told us they use
two port ranges for RTP, 7000-8000 and 1-2.
They use these ports. So there is nothing you have to do on Asterisk
side to handle this, as Asterisk's RTP ports are
The backtrace is not useable. Try to rebuild Asterisk with the Don't
Optimize Option (make menuconfig and the the build options)
regards
klaus
Edwin Lam wrote:
Philip A. Prindeville wrote:
On 03/08/2010 04:31 PM, Edwin Lam wrote:
hi folks.
i recently upgraded asterisk to 1.6.1.17(from 1.2)
Zoa wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent gatewaying.
The solution should be quite stable as long as the line quality is ok.
(Some tools for measuring the line quality are included in the release,
as
The libpri library is not found when loading chan_dahdi.so.
Try ldd /usr/local/lib/asterisk/chan_dahdi.so to see what dependencies
are missing.
Did you have installed libpri to a non-default directory? Then you have
to add the lcoation to the library path (something like /etc/ld.so.conf,
I am
Am 08.03.2010 11:10, schrieb mosbah.abdelkader:
Hello All,
MOS and R factor are the two QoS parameters used to estimate VoIP call
quality.
I have found that they are calculated from other metrics like jitter,
latency, packet loss,...etc. But, haven't found any formula or
arithmetic
Am 02.03.2010 07:26, schrieb Zhang Shukun:
hi, all
i want to realize more secure communication between asterisk sip end users.
so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption?
yes. But Asterisk does not support SRTP. Thus, only the SIP signaling is
encrypted, not the
Am 02.03.2010 08:50, schrieb Magnus Benngård:
Hi,
Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
problem to get outgoing calls to work but i have some problems with
incoming.
Did set srvlookup=yes in sip.conf. Sending all outgoing calls to
sip-corporate.tele2.se which
Am 13.02.2010 09:26, schrieb Olle E. Johansson:
12 feb 2010 kl. 16.43 skrev Klaus Darilion:
Am 11.02.2010 21:09, schrieb Olle E. Johansson:
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem
Am 11.02.2010 21:09, schrieb Olle E. Johansson:
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
[general]
rtpstart=3
rtpend=30100
Am 08.02.2010 21:15, schrieb Philippe Sultan:
Philippe, what exactly is a playback channel? Is it a pseudo participant
playing back the announcements?
Yes. Announcements are played to the conference members by creating a
channel of type 'Bridge' which streams the sound files.
thanks
Am 09.02.2010 15:35, schrieb David Backeberg:
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved my
problem.
Thus, any references to this new system are appreciated.
thanks
in the source code, I think that feature would
require a configuration file because the playback channel is not a per
user option.
Philippe
On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johanssono...@edvina.net wrote:
8 feb 2010 kl. 12.29 skrev Klaus Darilion:
Hi!
IIRC there was an announcement
E. Johanssono...@edvina.net wrote:
8 feb 2010 kl. 12.29 skrev Klaus Darilion:
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved
:
Klaus Darilion klaus.mailingli...@pernau.at
Date:
Tue, 10 Nov 2009 14:04:16 +0100
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi!
I am looking
Hi!
I am looking for a tool (application or webinterface) which shows me the
current status of an Asterisk server, e.g.:
- Status of the SIP peers (registered/offline)
- current incoming and outgoing calls
- start-time, numbers, some history
- history (calls stopped in the last 15
use pri debugging (pri debug span 1) to verify if the data sent on the
PRI line is correct! (e.g. type on number, ...)
verify with an incoming call and set the same format on outgoing calls.
regards
klaus
Jon Moore schrieb:
Hi list. I've googled around for this, and so far have come up
If the outgoing channel receives progress indication from the far end
(e.g. ISDN PROGRESS message or 183 response from an ITSP) then Asterisk
will relay the progress message. If there is no progress indication
received - that means that early media is not available - Asterisk does
not send 183
B.Masoud @ SH schrieb:
When Asterisk establish a call through an outbound trunk, Is there any
way I can know who hang up the call first? The caller or the party called?
you could use the 'g' option of the Dial command together with some
logic in the hangup extensions
regards
klaus
Hi!
Since 1.6, when using AEL, macros are implemented using Gosub(). Is
there workaround to have MACRO_EXTEN also in this case?
regards
Klaus
PS: I know I could use something like
context fromSip {
11 = myMacro(${EXTEN})
}
macro myMacro(MACRO_EXTEN) {
}
but isn't there some
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22
Danny Nicholas schrieb:
Sipregisterattempts would seem to be the simplest way to do this. It is 0
by default, changing it to 5 would stop the hacker after 5 tries.
wrong.
registerattempts wokrs the other way round - if Asterisk is the client
and registers to another SIP proxy.
regards
Steve Edwards schrieb:
On Mon, 5 Oct 2009, Klaus Darilion wrote:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
Don't you need another semi-colon after
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp
Do you have canreinvite=no in sip.conf? Maybe the variable is only set
if Asterisk is actually relaying RTP too.
regards
klaus
Asterisk User wrote:
Hi All,
While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use
it in my dialplan.
I had 2 sip extensions 555 and 666 and I
Benny Amorsen schrieb:
Jared Smith jsm...@digium.com writes:
Again, the emphasis on the dCAP exam is real-world knowledge of how to
build a simple small-business PBX with Asterisk. If you've used
Asterisk in a professional capacity, it should be very straightforward
to pass the practical
extension.ael:
if (0!=${MYVARIABLE}) {
...
}
or test for empty/unset variables, use:
${EXISTS()} or
${ISNULL()}
regards
klaus
michel freiha schrieb:
Hi all,
I need a goto If statement syntax that check if a variable is not null
then go to dialplan 1 else go to dialplan2
Regards
Lefteris Zafiris schrieb:
I have written a simple application for asterisk 1.6 that uses the Flite
tts engine to render text to speech.
Source is available here: http://zaf.github.com/Asterisk-Flite/
It works more or less like the festival app, can use cache etc.
Its only tested against
Olle E. Johansson schrieb:
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
Matt Riddell schrieb:
On 31/08/09 8:47 PM, Klaus Darilion wrote:
Olle E. Johansson schrieb:
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
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 (e.g. jitter, packet loss, reordering ...)?
Thanks
Hi Matt!
Matt Riddell schrieb:
On 27/08/09 9:24 PM, Klaus Darilion wrote:
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
John Todd wrote:
5) Any summary stats on RTP packet loss, etc? (from
CHANNEL(rtpqos,audio,all)) on channels?
I wonder how to retrieve those stats:
- after Dial()?
- during Dial()? (how?)
regards
klaus
___
-- Bandwidth and Colocation Provided by
Ishfaq Malik schrieb:
Hi
I know this is far from best practice but is it possible to authenticate
a sip peer on the IP address it's coming from so that it doesn't need to
use a UN and Pass?
Yes. That's exactly what type=peer is for.
regards
klaus
Hi!
I have the following setup:
PSTN--Asterisk-SIP--Asterisk
GW/LCR \ \ ...
\ \ ...
\ --SIP--Asterisk
\ ...
---Asterisk
The GW-Asterisk just does the
Are you using newest Asterisk versions? There were some similar problems
fixed recently:
https://issues.asterisk.org/view.php?id=13849
https://issues.asterisk.org/view.php?id=14239
https://issues.asterisk.org/view.php?id=14584
regards
klaus
Guillén Melo, Joaquin schrieb:
Hi, im trying to build
Miguel Molina schrieb:
Klaus Darilion escribió:
Hi!
I just found out that Asterisk (1.4) does not write CDRs if the incoming
call was not forwarded but handled internally without answering the call.
E.g.:
[from_pstn]
exten = 997,1,Answer()
exten = 997,2,Playback(tt-weasels)
exten
FYI: I checked the sources and Asterisk does write CDRs only if the call
in answered locally or forwarded to an outgoing channel.
Thus, as workaround I wrapped the extensions behind Dial(Local/...)
regards
klaus
Klaus Darilion schrieb:
Hi!
I just found out that Asterisk (1.4) does
Hi!
I just found out that Asterisk (1.4) does not write CDRs if the incoming
call was not forwarded but handled internally without answering the call.
E.g.:
[from_pstn]
exten = 997,1,Answer()
exten = 997,2,Playback(tt-weasels)
exten = 997,3,Hangup()
exten = 999,1,Playback(tt-weasels|noanswer)
1.6.0 is stable
1.6.1 is stable
1.6.2 is release candidate
See the files Changelog* and UPDATE* in this distributions for changes.
regards
klaus
Michael Cunningham schrieb:
Forgive me if this is a FAQ question but I didnt see anything on the
website
of forum spelling out the difference
Gordon Henderson schrieb:
Just been contacted by a UK Enum registrar looking for ITSPs to become
resellers of their Enum registration systems ...
Is anyone using Enum?
Yes.
Does anyone (other than cynical old me) think that Enum is a spammers best
friend?
I think ENUM will not cause
Xavier Cardil schrieb:
Hi, I've managed to get HYLAFAXT38MODEM-
ASTERISKCISCOAS5400 working, but when they are negotiating asterisk
drops a message telling Unknown RTP codec 96 received from gateway Do
somebody know how to fix it ?
Thank you !
[ TYPE: Control (4)
Jared Smith schrieb:
On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote:
I am searching for the description of the available dialstrin options
for the DAHDI channel (and also other channel types).
I am not looking for outdated voip-info links, but for the authoritative
source, e.g
Hi!
I am searching for the description of the available dialstrin options
for the DAHDI channel (and also other channel types).
I am not looking for outdated voip-info links, but for the authoritative
source, e.g. something like core show application Dial
Does such thing exists?
thanks
Klaus
Jay Ray schrieb:
Does asterisk support T38 passthrough now? What version onwards?
Since 1.4
ANy ideas on how to configure it for a host?
see sip.conf und search for 38 or udptl.
you should also look at udptl.conf and configure these ports in the firewall
regards
klaus
Steve Underwood schrieb:
Klaus Darilion wrote:
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
Benny Amorsen schrieb:
Klaus Darilion klaus.mailingli...@pernau.at writes:
Asterisk does not forward the 488 back to the caller, but hangs up the
callee's call leg. Further, the caller's call leg will not be hung up.
Is somebody aware of this problem and a fix?
This should be fixed
make sure to set canreinivte=yes for both peers
regards
klaus
Mindaugas Kezys schrieb:
Hello,
I want to send Media outside Asterisk server, e.g. between peers.
In CLI I see:
· [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging
SIP/5060-b7dc5218 and
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
if i understand you right you have one server (peer) where thousands of
devices are connected and every device is registered to asterisk, and so
every options packet will come from asterisk to this device, right?
If you have a sip
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI drivers.
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
INVITE
---200OK--
---200OK--
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
INVITE
Is there any options we need to enable in asterisk or grandstream phone?
I have already user transfer option 'Tt' in dialplan of this.
Please provide me some help.
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion
Max Alex wrote:
Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
Does it work with other phones? That means is it a Grandstream isue or a
general issue?
I have updated the latest firmware to the phone.
The phone is sending the *refer* to
Tzafrir Cohen schrieb:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our
Steve Underwood schrieb:
Hi Olivier,
Olivier wrote:
T.38 says that if the call starts in audio mode it is the called end
which should initiate a re-invite to change from audio to T.38. This
makes sense, as that is the end which has the best chance of figuring
out if a FAX
Configure emaildateformat in voicemail.conf.
I worked around the english weekdays by using numeric weekdays (see man
strftime)
emaildateformat=%d. %m. %Y um %H:%M Uhr
If you need the weekday in French you have to set the Linux Locale to
french. But this affects all parts of Asterisk where
Matt Riddell schrieb:
On 17/03/2009 9:10 a.m., Doug wrote:
So to make extension 201 in pickup group 1 just do:
asterisk -rx 'database put pickupgroup 201 1'
So this is a command line argument. Can this
be automated? Whenever we do a reload, can
this be stored?
The
answering myself ...
Klaus Darilion schrieb:
Hi!
AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE -
This is a bug which is fixed in Asterisk 1.4 branch, thus probably will
be fixed in 1.4.24
regards
klaus
but how should I handle the SUBSCRIBE in the context
ssmax wrote:
Hi all
i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp when
somebody dial +86 020 87654321 , the asterisk will get the call in
number by ${EXTEN} variable, but it can only get 87654321, no area
Jimmy Godbout wrote:
ssmax,
Use CALLERID(num) to get the number that was dialed.
CALLERID(num) is the calling number, not the called number
klaus
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There was already lots of discussion, e.g. google for
asterisk monitor nfs
or
asterisk monitor ramdisk
regards
klaus
Tarek Sawah wrote:
Hello,
a local prison contacted us regarding some calling card solution.
they need 4 E1s to serve 120 rooms in that prison.
we are planning on using 4
Hi!
Ist it possible with Asterisk to send SIP keep-alives with CRLF instead
of OPTIONS (qualify)? The OPTIONS are very noisy :-)
thanks
klaus
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