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,...
: application/sdpFrom:
"pavel"sip:[EMAIL PROTECTED]:5060;tag=as74cec9dbTo:
sip:[EMAIL PROTECTED];tag=3ef4af85-1112bVia: SIP/2.0/UDP
5.6.7.8:5060;branch=z9hG4bK5568aa34Content-Length: 168User-Agent:
Quintum/1.0.0
v=0o=Quintum 33034 31527 IN IP4
1.2.3.4s=VoipCallc=IN IP4 1.2.3.4t=0 0m=a
Hi there,
is there any possible way to forward UAC codec
capabilities to the PSTN gw w/ Asterisk?
Thanks,
Pavel
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Zot O'Connor wrote:
I am installing a used DG104S
I got it to ring from gnophone, but all I got was fast busies. so I
upgraded based on Pavel's link:
ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip
So I now have:
PROM Version: 3.0B22-DRUNTIME Version: 3.0B44-D
But
Zot O'Connor wrote:
I am getting (with older image):
RFC3389 support incomplete. Turn off on client if possible
How do I turn that off for the DG104s? Or if I can't how do I tweak
asterisk?
I see posts about ATA-186's having an audiomode, but the closet I came
to was inbanddtmf. I tried =0
Roy Sigurd Karlsbakk wrote:
hi
anyone around that wants to write MGCP transfersfers in *?
I need it, and I really don't know too much C...
roy
I have same needs,
there is good phone dlink dph-100m but * has no transfer in mgcp channel ...
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, extention.conf , log from CLI ...etc
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sip:[EMAIL PROTECTED]
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Eric Wieling wrote:
You must first purchase a license from the G723 patent holders. A
license costs about US$10,000.
I think the question was about what library do Mark use to compile g723
codec ...
first look is that the library is not from itu.org ...
On Tue, 2003-05-27 at 01:52,
', Identifier: '2', Endpoint: 'Protocol',
Version: 'Error or'
1 headers, 0 lines
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ICQ
#42433
But it seems not works,I always get conference call with 3 persons.
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on
'MGCP/aaln/[EMAIL PROTECTED]'
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Any ideas ?
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any ideas how to fix it?? or what I should check??
Thanks..
I had same problem - just rolled back to 0.2.1a
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Pavel Litvinenko.
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sip:[EMAIL PROTECTED]
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] with new mode: sendrecv on callid:
7d4b8e932149c6df
Posting Request:
MDCX 309 aaln/[EMAIL PROTECTED] MGCP 1.0
C: 7d4b8e932149c6df
What is the -1 'UNKNOWN' condition on channel ?
Is it correct mgcp packet ?
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Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http
When I connected over two mgcp channels and sending numerical
indication to cisco ata it seems hangup one channel (receving )
and generate 'fast busy' tone.
I hack chan_mgcp and my threewaycalling works ok!
But why indications are sent after I press hookflash on answering end?
--
Pavel
);
}
return 0;
}
works for me (tm)
You need cvs version, 0.4 does not work with flashhook messages at all.
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phone/fax +7(0732) 727172, http://www.comlink.ru
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issues with it.
Thanks,
Will
I was trying to get work 7940 MGCP with * - bad idea :) ... but it was
3.x version
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sip:[EMAIL PROTECTED
] )?
Tnx,
what incomming channel do you use, how is your * connected to the world ?
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sip:[EMAIL
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sip:[EMAIL PROTECTED]
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sip:[EMAIL
,
Pavel Litvinenko.
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*
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change the name of your gate from [192.168.0.5] to ip10
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ICQ
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Javier Rios wrote:
hello
you can help me with a problem
I have dlink DG-104S already and this registered in asterisk
but not to call... between in ports
you can help with an example the configuration me of
mgcp.conf
extensions.conf
; MGCP Configuration for Asterisk
;
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sip:[EMAIL PROTECTED]
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interface. Call transfer will be ma
naged through H.450.
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http
look at:
http://netphone.intracom.gr/english.htm
we have order this meanwhile for lab testing,
so I would be able to refer for about a month...
PJ
- Original Message -
From: Sean Cook
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Monday, December 06, 2004 12:19 AM
Subject:
so, better is to look to another phone, than surcharge cisco ;-)
PJ
- Original Message -
From: AST 386sx
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Monday, October 11, 2004 2:00 AM
Subject: Re: Re[2]: cisco ip 7905 legal ..
If you are going to buy it new. It should
, October 11, 2004 3:21
PM
Subject: Re: Re: Re[2]: cisco ip 7905
legal ..
Hello Pavel,well .. any GOOD propisition for the same
or lower price would be niceIP300 and ip500 are more expensif than
this one-- Best
regards,Danny
mailto:[EMAIL PROTECTED]belGOnet.com
a Euro-pictures
maybe Polycom is good phone, but Netphone seems to be cheaper (104, 114, 124 euro),
and (maybe) with more features:
in-line power (standard 802.3af cisco poe)
integrated switch (voice VLAN capable, learn via CDP!)
XML browser (callmanager compatible)
corporate phonebook!
SIP, h323, sccp!
you know where I could buy one of these
phones from?I'm based in the UK, but would like to play around with
one. Iemailed Intracom but got no response.On
Thu, 14 Oct 2004 22:59:33 +0200, Pavel Jezek [EMAIL PROTECTED] wrote: maybe
Polycom is good phone, but Netphone seems to be cheaper
you can use vovida's open g729 sample code, look
to:
http://www.voiceage.com/codecsite/openinit_g729.php
PJ
- Original Message -
From:
Matthew
Boehm
Newsgroups:
gmane.comp.telephony.pbx.asterisk.user
Sent: Tuesday, October 19, 2004 6:03
PM
Subject: Re: GSM to
Daniel ANDRE wrote:
Mariam,
I have added these lines and still no transfert menu on my IP10S
they must be aded before line = aaln/1
transfer works by flash event
Regards,
Daniel
Pavel Litvinenko wrote:
Marian Danisek wrote:
Daniel ANDRE wrote:
I have the MGCP Firmware and call
list
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Yes, I do :) where can I get it ?
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ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED
/listinfo/asterisk-users
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mailing list
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Andrew Thompson wrote:
- Original Message -
From: Pavel Litvinenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 8:42 AM
Subject: Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
bam wrote:
I've read through the archives and have picked up
Anton Yurchenko wrote:
Hello,
I`m try to make the attended transfer work Dlink DG-104S via FLASH,
when somebody calls my phone I pickup and press flash to get a second
line to call another extension. When I press flash I hear no dialtone,
and only a long and then small beep. When I try to
Pavel Litvinenko wrote:
Anton Yurchenko wrote:
Pavel Litvinenko wrote:
Anton Yurchenko wrote:
Hello,
I`m try to make the attended transfer work Dlink DG-104S via FLASH,
when somebody calls my phone I pickup and press flash to get a
second line to call another extension. When I press flash
I thing, that configuring nat device/firewall at consumer site isn't
always possible, thus simplest (but not optimal) way is to configure
phone in sip.conf as nat=yes canreinvite=no, this should work in most
cases even if multiple phones are behind same nat, like adsl router.
disadvatage is,
maybe usefull for displaying CALLED party name when dialing
I'm remember, that this feature was planned to add to asterisk, any
progress?
PJ
New and Changed Information
Release 8.0(0) includes the following new and enhanced features:
•Remote-Party ID support has been added for
Hello, sometimes I'm experiencing autocongest error due slow response,
anyone knows, what this means?
Second or third attempt after that happens pass successfully...
this happens ever in fastethernet lan, so no problem with lag in wan
environment,
I'm using idefisk 1.32 on client side (winxp
I have also issues with jitter over wan (cdma),
I'm trying to debug how dejitter buffer is working (using iax2 jb
debug), but nothing happens/no debug output on asterisk console :-(
is any way how to monitor iax jitter buffer? thx
PJ
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Hello, I would like apply different call rate (tarif) per outgoing
number (or group of phones, based on prefixes),
I'm playing with astpp, but seems, that this feature isn't available here,
can you recommend any other open-source billing (A2billing, AstBill?),
that this can do?
thank you!
PJ
Ron, keep in mind, that yoy mix parameters for new and old iax
jitterbuffer implementation, these:
dropcount=2
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
are ae valid only for _old_ implementation, and I thing, that asterisk
1.2 use new iax buffer by
same for me ci$co router via chan_h323 (a have vad disabled on voip
dial-peer on router),
but I'm ignoring this notice messages from asterisk console and logs,
because calls are not disturbed ;-)
PJ
FaberK wrote:
Hi guys,
I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture
g729A is light g729 version, good compromise between cpu requirement
and quality (little worse than original g729)
g729annexB have built in VAD and comfort noise generator
g729AannexB is both above
I thing, that streams from all this codecs are mutually compatible, and
asterisk supports only
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
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AstriCon 2008 -
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 know
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: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users
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
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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 primary intended use is in conjunction
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
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_sip,
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 cpu and lock any
access to computer
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
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-
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
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
but if I try to use gosub in extensions.ael, ael
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 discussion on asterisk IRC, where people said, that
macros will be depracated, I tried
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:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] 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 commercial
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)
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
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?
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
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
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
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
:
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
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
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
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,
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
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 progress
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 have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...
device
deviceProtocolSIP/deviceProtocol
sshUserIdadmin/sshUserId
sshPasswordadmin/sshPassword
devicePool
dateTimeSetting
any idea, how to do something like this, but in correct/functional form?
;-)
Set(foo=System(echo ${EXTEN} | tr [:upper:] [:lower:]))
${EXTEN} is SomeStrinG
${foo} output should bee somestring
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another reason, why better is to completely avoid ci$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
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
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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
it is in doc/ directory
asterisk-conf.txt
Tomislav Parčina wrote:
Why there is no asterisk.conf.sample file?
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imho, ci$co doesn't support anything other than callmanager as signaling
server :-(
Peter Mitchell wrote:
79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.
Whats the non callmanager - SIP licence number for 79X1 ?
and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any idea, how to do something like this, but in correct/functional
form? ;-)
Set(foo
I think, sip server even doesn't know, that user picks up handset,
maybe with skinny or mgcp phone should it work because this phones are
controled by signaling server
PJ
chester c young wrote:
On a SIP phone is it possible to enter the dialplan when the user
picks up the phone without
something like (AEL syntax):
if (${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.
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
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 ${EXTEN});
}
-- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/bill-gw-10,
WaitExten is useless in this case, because it's waits for user input,
but we are talking about executing diaplan when entering 'h' extension,
ie. after user hangs up phone...
and seems, something strange with processing wait() app in processiong
'h' extension in diaplan - timeout specified is
Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for bussines use :-\
snoms are probably also good, but functionality/price ratio is, imho,
better for thomson or
for massive deployment phone provisioning/fw updating through web
interface is not optimal,
best way is via config files/templates periodicaly downloaded from
central tftp/http server...
PJ
MF wrote:
Best and easiest provisioning I´ve found imho is Snom, great web
interfase , followed by
ci$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
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...
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
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
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
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