23.04.2019 0:27, Joshua C. Colp wrote:
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:
Tried already.
"line" is good, but not perfect.
Every time I restart asterisk, it will generate new random string for ";line=".
So, every time I restart asterisk, registrar (Server1
Hi,
Thank for your answer.
22.04.2019 23:47, Joshua C. Colp пишет:
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
Registration is OK.
Server2 outgoing calls are OK.
INVITE, unauthorized, INVITE with password, OK, RINGING,...
Troub
On 07/20/2013 09:00 AM, Code Lover wrote:
[introutingB]
exten => s,n,Dial(SIP/voipprovider)
Try adding a '1' priority to the inroutingB block:
[introutingB]
exten => s,1,Dial(SIP/voipprovider)
--
_
-- Bandwidth and Colocatio
Any suggestions? Or I should address it to asterisk-dev maillist as feature
request.
Regards,
Pavel
>Hi David,
>
>For sure I will ask them but I think Asterisk should be able to handle this
>case because it doesn't matter if it is adhearsion or something else. If it
>is not
is is the normal behaviour of async agi
Regards,
Pavel
>Pavel,
>This may be due to the asynchronous nature of Async AGI. The
AGI(agi:async) command will continue to run until the channel is hung up,
the 'asyncagi break' command is sent over AMI, or some sort of fatal error
is >e
Danny as I mentioned in the first message the dialplan execution stops
after AGI(agi:async). So the provided example doesn't help.
Regards,
Pavel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
N
Hi Danny,
I am running it asynchronously because adhearsion needs it.
Regards,
Pavel
Date: Wed, 5 Sep 2012 11:28:46 -0500
From: "Danny Nicholas"
Subject: Re: [asterisk-users] Async AGI
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Mess
=> _X.,2,Answer
exten => _X.,3,Playback(some-message)
exten => _X.,4,Hangup
Regards,
Pavel
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_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory we
C. Savinovich wrote:
> Can somebody please give a pointer to a complete neophyte (like me) on
> text messaging, what product can I use to send and automatic text message to
> a cell phone from within the asterisk dialplan? (the part of the dialplan I
> have down, the part of the text message no
Atis Lezdins wrote:
> On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek <[EMAIL PROTECTED]> wrote:
>
>> Steve Murphy wrote:
>>
>>> On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
>>>
>>>
>>>> Atis Lezdins wrote:
>&
Vieri wrote:
> Hi,
>
> Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user
> wants to pick up a call within his/her pickup group, *8 must be dialed (or
> whatever you define in features.conf).
>
> However, these users were used to another behavior when they had a commercia
Steve Murphy wrote:
> On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
>
>> Atis Lezdins wrote:
>>
>>> On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek <[EMAIL PROTECTED]> wrote:
>>>
>>>
>>>> Hi, according to discu
Atis Lezdins wrote:
> On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek <[EMAIL PROTECTED]> wrote:
>
>> Hi, according to discussion on asterisk IRC, where people said, that
>> macros will be depracated, I tried to migrate from macros to contexts
>> and Gosub
&
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosu
you should issue 'sip show peers' command to see, if your phones are
available,
put 'qualify=yes' in your sip.conf
'sip show registry' command is usefull to see if your _asterisk_ is
registered to some another sip server, eg. voip provider..
PJ
David Boyd wrote:
> -Original Message-
>
Nhadie wrote:
> I disabled logging of NOTIFY on the CLI and it does not show anymore,
> however CPU is still very high, latency as well goes up when it is
> trying to poke my phone here, my phone(SPA942) also keeps on rebooting
>
> is there a way to increase the time of sending the qualify? TIA
Tilghman Lesher wrote:
> On Wednesday 06 August 2008 04:09:13 Pavel Jezek wrote:
>
>> A week ago, I tried give realtime priority to asterisk proces using -p
>> switch,
>> asterisk was running inside astcanary,
>> but yestarday asterisk probably starts eating all
Steve Murphy wrote:
> Hello--
>
> Why do I target chan_sip for so much effort? Because,
> it seems to me, chan_sip is probably the most used channel
> driver in the asterisk community!! (and, of course,
> the zap/dahdi driver, is also pretty popular)
>
> I haven't had time to follow up on chan
A week ago, I tried give realtime priority to asterisk proces using -p
switch,
asterisk was running inside astcanary,
but yestarday asterisk probably starts eating all cpu and lock any
access to computer, only ping was possible,
so, anybody have experience, that ascanary process does really work
Tilghman Lesher wrote:
> On Sunday 27 July 2008 13:28:00 Pavel Jezek wrote:
>
>> Hi, can somebody explain how to use this func/apps in asterisk?
>> I tried to find some examples on mailinglists or wiki, however without
>> success. thanks
>>
>
> The pr
try put calls into groups using GROUP() function and check call limit
with GROUP_COUNT()
voip crazy wrote:
> Hello list,
>
> How could I limit the outgoing calls for one trunks easily?
>
> Thanks
>
> VoipCrazy
>
> ___
> -- Bandwidth and Colocation Pro
New in Asterisk 1.6
ronald ramos wrote:
> hi,
>
> thanks for your reply. is dialgroup already available in asterisk 1.4?
> i'm currently using 1.4.21.
>
> regards,
> ron
>
> --- On Mon, 7/28/08, Pavel Jezek <[EMAIL PROTECTED]> wrote:
> From: Pave
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
> Hi,
>
> Would just like to kn
Hi, can somebody explain how to use this func/apps in asterisk?
I tried to find some examples on mailinglists or wiki, however without
success. thanks
PJ
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - Septembe
after recompilling asterisk (trunk-r75109) after system (mandriva
cooker) update (new glibc 2.6, gcc 4.2.1),
sound starts very choppy, when codec translation is performed,
if translation isn't needed, it sounds OK
any idea? until update, everything worked fine.
I'm using ztdummy as clock source.
you should turn on sip debug on asterisk and median and see, if sip/180
ringing messagess are propagated through mediant to avaya,
avaya should react to sip/180 ringing with generating ringback to
calling phone...
sip/183 is progress message, in this case is audio path "open" to
playback progres
If you flash new sip flash firmware into 7941 look at tftp log, you will
see, that after firmware flashing and phone reboot, it will download and
flash localization files in next flashing cycle,
if you copy this files from callmanager tftp dir to your tftp server it
will work.
before flasing loc
somebody knows, what this mean, or how to avoid this messages?
I have clock synchronized on asterisk server using ntpd.
Internal RTCP NTP clock skew detected: lsr=4103127456, now=4103296271,
dlsr=168820 (2:575ms), diff=5
Internal RTCP NTP clock skew detected: lsr=4103522652, now=4103656826,
dls
some work has been done here:
http://bugs.digium.com/view.php?id=4825
but seems to be quite death and probably not directly applicable to
current asterisk src :'(
SIP wrote:
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by on
to provide me
with a copy of your SEP.cnf.xml file and whatever other files the
phone uses so I can ensure that its not something else? Thanks.
Eric
On Fri, 2007-06-01 at 15:07 -0500, Greg Oliver wrote:
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
we are using 7941 with sip
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-)
Eric Lubow wrote:
All,
I am having a lot of trouble with the Cisco 7961G phones. I have
managed to get them up and running with Asterisk to the point where I
can get incoming calls and make outgoing calls. The problem is wh
recently added support (with bug) for SendURL for SIP channel causes
problem with nokia phones, as I reported in
http://bugs.digium.com/view.php?id=9821
it was quickly resolved,
but because I can't find any RFC what it is doing/how to use it, I would
like to ask here,
if someone using this featu
do you have also compiled latest svn-trunk zaptel?
Iban Lopetegi Zinkunegi wrote:
Hi all!!
I have downloaded the asterisk from svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the
asterisk 1.4 subversion). I also downloaded the patch for cellphone
and make it work fi
spa-922/942 has backlighted display, inline power (PoE), internal switch,
audio gain/attenuation can be tunned,
works great in bussines environment (voice vlan negotiation through cdp
from ci$co switch), solid design, robust chassis
lack of features like programable buttons for pickup or busy la
callmanager can also be running in ios firmware in router (callmanager
express), with near all funcionality as server version...
Adam KOSA wrote:
Antonopoulos Angelos wrote:
Thanks for your help..But i dont know yet whether is CCM embeded on
cisco 2851 or it is an extra element?
Practical
hello, is there any equivalent, that is currently usefull, if I have
some iax connections with jitter spikes and another with minimal jitter?
for my jittery connections, I don't like to shrink jitter buffer too
fast, because another jitter spike can occur again and small jb can't
cover it.
as I
any recommendation for usable softphone for windows with srtp support?
I'm using CounterPath eyeBeam, but seems, that doesn't support secure
rtp at all.
... and minisip seems to be quite death project, without activity.
PJ
marek cervenka wrote:
On Fri, 2007-03-23 at 16:12 +0100, marek cervenk
yed nor could it
phone out or receive any calls...
Anyone able to share some snippets of their skinny.conf?
I just used the examples and modified the MAC line and
extension line config...but seems something else is
missing...
cheers
rick
Pavel Jezek schrieb:
If you have 7970 right configure
If you have 7970 right configured to point to asterisk server, you
should be able to see some skinny debug on console, or look what report
"skinny show devices"
I haven't any 7970, so can't help so much, I'm using only 7920 wifi
phone with chan_skinny and 1.4trunk, it's usable, basic functionali
last chan_sccp was released a year ago,
Sergio, main developer, gone away
minimal activity in forum,
chan_sccp.org, unoficial chan_sccp site, is for sale
this are reasons, why I also considering chan_sccp as death project.
Bill Hackensack wrote:
chan_sccp is far from dead and it works with
isn't maintained as good as chan_h323,
if you will have some issues with chan_h323, you can report through
digium bugtrack and reply come quite quickly, not so with chan_ooh323
thats my reasons, why I can recommend original chan_h323.
PJ
Thiago Maluf wrote:
Hi Users, Administrators and P
as I know, ooh323 is external project from objective systems,
anyway, for 1.4 I prefer chan_h323 from asterisk tree.
Thiago Maluf wrote:
Now, the H323 Channels is updated and your bugs fixed.
But Digium still develop your OOH323 Channel. My question is why?
What is the better in Asterisk 1.4.x
you should separate to two lines, like...
exten => _366[5-9]X,...
exten => 36700,...
Hall, Eric M. wrote:
D
Not sure why this works
exten => _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten => _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context
Olivier wrote:
That was exactly what I meant.
Your setup is :
Nortel --- Cisco --- Asterisk
What I was thinking about is:
Nortel --- Asterisk1 --Asterisk2
In previous case, your are using Cisco's QSIG features.
In the latter one, you could use Asterisk QSIG fea
Brian Capouch wrote:
But the included comments say, "The user part of a type=friend call
will still be affected by the call limit"
Those seem to be in conflict, but perhaps it's just my parser :-)
Could someone clueful explain?
I interpret this that asterisk _internally_ still counting
Olle E Johansson wrote:
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
quite OT:
do you have some info, about what platforms supports qsig decode? I
found, that first supported in 12.4.(9)T, but I don't know, if only on
28xx or also in older routers (like 3660) with NM-HDV-E1...
and what is the name for qsig decode feature? I compare features in two
ci$co firmware
interesting!
so it means, that you can now see caller id names between sip phones
connected to asterisk and phones connected to pbx?
PJ
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and
Nortel TX-1. I had problems with
I think, this can be solved using phone autoanswer feature, look at wiki...
exten => s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten => s,2,Dial(SIP/myphone)
Paradise Dove wrote:
hi,
is there anyway to Answer() the caller channel after the called number
pickedup the phone.
when
context 'default' has not any special, it's context, that will be used
if your peers/users definition doesn't contain any specific context
if you have permited 'anonymous' calls to your asterisk, i.e
allowguest=yes, unautenticated calls (calls, that will not match any
specific user in sip.conf)
Is there any way, how to detect, what party starts touch monitor
recording? is some variable set?
I would like to deliver recorded file after call hangup to that user
using some shell script.
PJ
___
--Bandwidth and Colocation provided by Easynews.com -
did you use correct context to pickup external call?
if you simply pickup without context, it will try to pickup ringing line
in "@from-internal" context, from you example...
PJ
nik600 wrote:
I am trying to configure the pickup.
This is my dialplan:
exten => _57.,1,Pickup(${EXTEN:2})
So, w
if you can't use asterisk for recording ;-)
you can try zoom-int callrec, this works listening on switch span port
to record calls...
but it's not free app
Cory Andrews wrote:
Apologies in advance as this is not directly Asterisk related, however I
thought I might be able to leverage the ex
you just post only call forward activation part of dialplan,
but you must also make dialplan part, that reflect, how is set this
callforward mark,
ie. if callforward is set, dial that number, if not, dial peer...
Dominik Zalewski wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding
Jens Vagelpohl wrote:
I have two APs (Apple AirPorts) sending on the _same_ channel.
Handover works perfect with no discernible loss of connectivity or
audio using a Siemens SL75. The handover cannot even be noticed.
as I know, best practice says, that neighboring AP should use _non
overl
some howto configuration for asterisk controlling ci$co router (pri/qsig
ports especially) using mgcp interests me too... ;-)
Yehavi Bourvine +972-8-9489444 wrote
I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is
a "voice bundle" of 2,811 + E1 + PDLM card. Note
new ci$co phones are compliant with 802.3af, but are incompatible with
asterisk ;-)
.cnf.xml config files are undocumented, remote phone management (eg.
restart) is very difficult, if you are not use callmanager
personaly can't recommend new ci$co phones, nor obsolete models, like
7912/40/60...
ci$co phones are definitively not good choice if you would like to use
with anything other than callmanager as signaling server (especially
true for new models 7911/41/61/70)
Michelle Dupuis wrote:
We used Aastra's for a good while, but gave up on them (and switched
to Cisco). Aastra's seem
for massive deployment phone provisioning/fw updating through web
interface is not optimal,
best way is via config files/templates periodicaly downloaded from
central tftp/http server...
PJ
MF wrote:
Best and easiest provisioning I´ve found imho is Snom, great web
interfase , followed by Pol
Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for bussines use :-\
snoms are probably also good, but functionality/price ratio is, imho,
better for thomson or li
WaitExten is useless in this case, because it's waits for user input,
but we are talking about executing diaplan when entering 'h' extension,
ie. after user hangs up phone...
and seems, something strange with processing wait() app in processiong
'h' extension in diaplan - timeout specified is ig
I can confirm,
commands after Wait() are never executed in 'h' extension
and wait seconds argument in wait() is completely ignored
it's bug or "feature"? ;-)
h => {
NoOP(before ${EXTEN});
Wait(5);
NoOP(after ${EXTEN});
}
-- Executing [EMAIL PROTECTED]:1] NoOp("IAX2/bill-gw-10",
switch is layer two device and transparent to communication asterisk to
phone
Michael Welter wrote:
Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems
causing "retries exceeded" in Asterisk?
Thanks
___
--Bandwidth and Colocation
something like (AEL syntax):
if (${DB_EXISTS(cidname/${CALLERID(num)})})
CALLERID(name)="${DB(cidname/${CALLERID(num)})";
Derek Whitten wrote:
WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is
deprecated.
Please use ${DB(cidname/${CALLERID(num)})} instead.
[W
I think, sip server even doesn't know, that user picks up handset,
maybe with skinny or mgcp phone should it work because this phones are
controled by signaling server
PJ
chester c young wrote:
On a SIP phone is it possible to enter the dialplan when the user
picks up the phone without h
l a script and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any idea, how to do something like this, but in correct/functional
form? ;-)
imho, ci$co doesn't support anything other than callmanager as signaling
server :-(
Peter Mitchell wrote:
79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.
Whats the non callmanager - SIP licence number for 79X1 ?
___
it is in doc/ directory
asterisk-conf.txt
Tomislav Parčina wrote:
Why there is no asterisk.conf.sample file?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
-
I think, ci$co phones can not be even purchased without licence...
btw, what is your reason, to buy ci$co phones, when known issues exist
with this phones, if working with anything other than callmanager? :-\
PJ
Peter Mitchell wrote:
I've got a question regarding Cisco IP Phones and licenci
asterisk.conf
[options]
verbose = 3 ; Verbosity level for
logging (-v)
Neil Tancock wrote:
Hi, how do I get Asterisk to start in very verbose mode every time it
boots?
Neil
___
--Bandwidth and Colocation pr
another reason, why better is to completely avoid ci$co phones when used
with anything other than callmanager ;-)
Yehavi Bourvine +972-8-9489444 wrote:
The users want the transfer softkey on the screen while on a call.
Currently it is acessable via the more softkey.
I've asked Cisco whe
any idea, how to do something like this, but in correct/functional form?
;-)
Set(foo=System(echo "${EXTEN}" | tr "[:upper:]" "[:lower:]"))
${EXTEN} is "SomeStrinG"
${foo} output should bee "somestring"
___
--Bandwidth and Colocation provided by Easyn
you have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...
SIP
admin
admin
D-M-Y
Central Europe Standard/Daylight Time
ntpserver
I think this is wrong use,
for conferencing, you must dial meetme application and put all users
into one conference room...
PJ
nik600 wrote:
Hi, can i set up a similar dialplan?
Suppose that extension make the caller to join a conference room.
extension => 3,1,Dial(SIP/200,SIP/[EMAIL P
I prefer h323 included in asterisk tree,
I have caller id issues with ooh323 and nobody answer to bugreports
oh323 from inaccessible network is unmaintained/death project,
incompatible with asterisk 1.4.
PJ
Michel wrote:
Hello,
I need your advice about H323 and asterisk! ;) Which one do yo
but keep in mind, that jb for sip (generic jitterbuffer) is implemented
differently, than iax, so it works only for SIP-SIP calls, or SIP-ZAP
and, curious, eg. for SIP->ZAP call must be activated for (outgoing) ZAP
channel :-\
yusuf wrote:
[EMAIL PROTECTED] wrote:
In iax.conf there is optio
linkys is definitively one of the most noisy switch!
it must be placed far away from people :-)
Noah Miller wrote:
I have an upcoming install which places the switch close to some
employees
in a quiet work environment. Can anyone recommend a quiet 24 port POE
switch? The Linksys SRW224P be
this limitation still persist in 1.4 or trunk? if yes, it should be
really improved...
imagine, even small companies have more than 32 offices, i.e.
callgroups, where phone call pickups are needed.
John Harragin wrote:
callgroups & pickupgroups greater than 31 are not working for sip calls
w
we probably need to ask in dev- list, because seems that only developers
knows, how to use/test SLA feature ;-)
Anthony Kava wrote:
Greetings,
Back in September someone asked about documentation for the new SLA feature
in 1.4, however they received no replies. I thought I might ask the same
no, ipsec headers add much more traffic overhead even than small voice
rtp packets bears (using low bitrate codec).
this ipsec overhead is not too crucial when ecapsulating relatively big
data packet
Benny Amorsen wrote:
"PJ" == Pavel Jezek <[EMAIL PROTECTED]> writes:
some idea, how to make BLF working on ci$co 7961 (sip)?
Steve Langstaff wrote:
I've not used the Cisco kit for this, but you might try adding 'hints'
to your agent extensions, and then defining a BLF button to subscribe to
this.
e.g. If you have an agent with ID 1001, add this to extensions.c
1) why you Answer() before Dial()
2) try Dial(SIP/user) instead of Dial(SIP/[EMAIL PROTECTED]) asterisk
knows, what IP has peer (sip show peers)
3) try call echo() test aplication from callmanager phone
nik600 wrote:
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
in h323.co
in h323.conf you have still:
disallow=all
allow=all
try change to:
disallow=all
allow=alaw
nik600 wrote:
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
probably you haven't g729 installed in asterisk, use g711 instead, put
this in h323.conf and in callmanager plac
I think, callmanager needs media termination point (mtp) for sip trunk,
so rtp stream will always go through callmanager...
JR Richardson wrote:
Hi All,
I haven't started sip traces or debug yet, but was wondering what the deal
is with the CCM and reinvite, why it doesn't work with Asteri
probably you haven't g729 installed in asterisk, use g711 instead, put
this in h323.conf and in callmanager place asterisdk gateway in region
that will use g711...
disallow=all
allow=alaw
alternatively you can find g729 codecs binaries here:
http://kvin.lv/pub/Linux/Asterisk/
nik600 wrote:
H
]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 4:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to define a secure trunk
as I know, only preliminary support:
0005413: [patch] Secure RTP (SRTP)
http
as I know, only preliminary support:
0005413: [patch] Secure RTP (SRTP)
http://bugs.digium.com/view.php?id=5413
Joao Pereira wrote:
Can I do the encrypted trunk in SIP? Does Asterisk supports it?
Thanks
Joao Pereira
Pavel Jezek wrote:
http://www.voip-info.org/wiki/view/IAX+encryption
I think, Nokia E60/61/70 currently supports 802.1x
Joao Pereira wrote:
Do you know if it has 802.1x authentication as it is defined in
EDUroam ( http://www.eduroam.org/ ) ?
I never found a WiFi phone working with 802.1x I tested ZyXel
Prestige 2000 but the sound was bad and it doesnt supp
http://www.voip-info.org/wiki/view/IAX+encryption
Joao Pereira wrote:
Hello
I would like to define a trunk from my Asterisk to a VoIP provider,
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls
aren't intercep
5.txt
nik600 wrote:
Ok thanks, do you think that it isn't possible to do that
automatically from asterisk?
On 12/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I think, that adhoc conferencing isn't possible in this way, instead you
should use meetme, ie.:
skinny user calls to user
CDP has nothing to do with inline power, it is L2 proprietary protocol
for negotiation of voice vlan between phone and switch,
so you don't need to set what vlan number phone should use for voice and
what is for connected pc data.
if you disable cdp on switch, phone will still working, except you
Matt Gibson wrote:
Hi Pavel,
I tried to implicitly set qualify=no for the sip user, but am still
seeing the "registering" icon for like 10 minutes on the screen of the
7970. It is actually registering, just the phone doesn't think it is.
The phones always stay with a littl
I'm using 8.2.1 in 7961, it working fine, registration is OK, except I
must disable qualify in asterisk (phone doesn't respond to qualify "pings"),
one anoying bug removed is not displaying IP address of sip server
(asterisk) in caller id,
also same issue with needing rename jar*.sbn file on tft
-= Info about application 'MixMonitor' =-
[Synopsis]
Record a call and mix the audio during the recording
[Description]
MixMonitor(.[|[|]])
Records the audio on the current channel to the specified file.
If the filename is an absolute path, uses that path, otherwise
creates the file in the co
I think, that adhoc conferencing isn't possible in this way, instead you
should use meetme, ie.:
skinny user calls to user A and transfer his to meetme number
skinny user calls to user B and transfer his to meetme number
skinny user calls to meetme number
all three speech in conference...
ni
nobody knows, how jitterbuffer actually working when asterisk doing
protocol translation? i.e. sip-iax, skinny-iax...
how current two jb implementations (generic rtp & iax jb) working together?
PJ
Pavel Jezek wrote:
so that, jitterbuffer should be enabled & forced on sip and iax
ch
so that, jitterbuffer should be enabled & forced on sip and iax channel
on asterisk (because UAs have no knowledge about jitter on opposite
link), from first example?
UA(sip)--->OpenSER--> Asterisk--> UA(IAX2)
Steven wrote:
Nothing is end to end in this case.
It is two sep
how can protocol translation affect jitter propagation to both voip ends
(UAs) for dejjiterring? because iax doesn't use RTP for voice stream, it
can be issue (?)
PJ
David Thomas wrote:
Yes, as long as Asterisk is in between the two, it can perform the
protocol translation.
regards
David
On
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel Hiver wrote:
[EMAIL PROTECTED] a écrit :
Hi all,
I'm looking at some suggestions from you techies out there.
Let me explain my scenario. Im a reseller to callshops.
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