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
--
___
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
--
__
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 06
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 pu
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 me
, Klaus Darilion wrote:
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 trav
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
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 o
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 (ori
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 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 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 TAP
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 (com
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?
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,
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 arou
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 per
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 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 t
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 19:31, schrieb Matt Watson:
>
>
> On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion
> 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 th
Am 18.03.2010 05:11, schrieb Olivier:
>
>
> 2010/3/17 Klaus Darilion <mailto:klaus.mailingli...@pernau.at>>
>
>
>
> Am 17.03.2010 10:40, schrieb Peter den Hartog:
> > Hello,
> >
> > I was wondering if the following was possible:
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 instal
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
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 ever
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 DU
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,
>> e
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 t
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 mac
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
>>
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
lik
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
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-corporat
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
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 diffe
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 wel
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
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
> ari
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 a
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.tel
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 th
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 11.02.2
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 allocati
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
>rtp
Am 09.02.2010 15:35, schrieb David Backeberg:
> On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
> 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
>> i
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.
>
>> tha
l is not a per
> user option.
>
> Philippe
>
> On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson wrote:
>>
>> 8 feb 2010 kl. 12.29 skrev Klaus Darilion:
>>
>>> Hi!
>>>
>>> IIRC there was an announcement some time ago that it is possible now
7;t have the playback channel speak French,
> from what I've read 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. Johansson wrot
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
klaus
> [asterisk-users] looking for an Asterisk supervision (status viewer) tool
>> From:
>> Klaus Darilion
>> Date:
>> Tue, 10 Nov 2009 14:04:16 +0100
>> To:
>> Asterisk Users Mailing List - Non-Commercial Discussion
>>
>>
>> To:
>> Asteris
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 short
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 minutes,
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
_
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
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
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;
>>
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!
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
kl
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
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(); }
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 ${RTPAUDIOQOS} and tried to use
> it in my dialplan.
> I had 2 sip extensions 555 and 666 and
extension.ael:
if (0!=${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
>
> Reg
Benny Amorsen schrieb:
> Jared Smith 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 portion
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
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 a
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
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
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 Aster
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
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
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
Hi!
I have the following setup:
PSTN<-->Asterisk<-SIP-->Asterisk
GW/LCR \ \ ...
\ \ ...
\ --SIP-->Asterisk
\ ...
--->Asterisk
The GW-Asterisk just doe
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)
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
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 be
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|noansw
Xavier Cardil schrieb:
> Hi, I've managed to get HYLAFAX>T38MODEM->
> ASTERISK>CISCOAS5400 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
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 c
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
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
Klau
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
>
>
>
> ---
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.
http://lists.digium.com/pipermail/asterisk-users/2008-April
Benny Amorsen schrieb:
> Stefan Schmidt 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 ro
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-b7dc
Benny Amorsen schrieb:
> Klaus Darilion 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?
>
Steve Underwood schrieb:
> Klaus Darilion wrote:
>> Atis Lezdins schrieb:
>>
>>> On Mon, Jun 8, 2009 at 2:06 PM, Klaus
>>> Darilion wrote:
>>>
>>>> Hi!
>>>>
>>>> I have the following problem with Asterisk
Atis Lezdins schrieb:
> On Mon, Jun 8, 2009 at 2:06 PM, Klaus
> Darilion wrote:
>> Hi!
>>
>> I have the following problem with Asterisk 1.4.23:
>>
>>
>> ATA w/ T.38 Asterisk ATA w/o T.38
>> INVITE---
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--
ACK--
>>
> Content-Length: 0
>
>
> <>
>
> 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
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
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 fo
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 times
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
>> ou
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 st
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 usin
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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asterisk-users mailing
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
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
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