sangoma cards do not use dahdi...
13.5.2011 v 17:16, satish patel satish...@hotmail.com:
Thank you so much!! I found following (res_timing_timerfd.so in USE). But we
have asterisk dahdi install and sangoma A102D pri card configured. Do you
think i should use res_timing_dahdi.so ?
Matthew,
ok, but is realy possible change the dsp code in the Asterisk? Guys around
The OpenPBX change the dsp to Steve's spandsp and has the native T38 support
now.
Tomas Urbanek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
compile kernel without usb support or unload usb modules
turby
ps
your tdm card don't share the irq, your network card share the irq...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, December 14, 2006 5:13 PM
To: Asterisk Users
joao,
you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.
turby
-Original Message-
From: [EMAIL PROTECTED]
[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
right, but who have production and tested code of application level
encryption for SIP and IAX for SECURE(!) trunks?
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 6:15 PM
To: Asterisk Users Mailing
check cdr_mysql.conf for userfield=1
turby @ www.canistec.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tristan
Sent: Wednesday, June 07, 2006 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Set(CDR
convert the moh sounfile to pcm or sln
save the file to
/var/lib/asterisk/moh/default
set the musiconhold.conf
[default]mode=filesdirectory=/var/lib/asterisk/moh/default
turby@ www.canistec.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
ReinaSent: Wednesday
use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database)
astdb=chan2ext/SIP/grandstream1=1234 is only variable
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Thursday, June 01, 2006 9:39 AM
To: asterisk
,Goto(dialthru,s,2)
exten =
_X.,1,Dial(TRUNK/${EXTEN})
...
turby
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change context to context=remote in [general] in sip.conf
you missing registration of peer :)
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of btb
Sent: Thursday, February 23, 2006 4:10 AM
To: Asterisk Non-Commercial Discussion Users Mailing List
[dial]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,2,Congestion
exten = _X.,102,Busy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan Alberti
Sent: Thursday, February 23, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
yes, with last patch works well. thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adolfo R.
Brandes
Sent: Thursday, February 16, 2006 10:11 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: asterisk t.38 pass
turby wrote
this is not usefull for public enviroment. clients behind
nat does not work...
turby
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitin
GuptaSent: Tuesday, February 14, 2006 10:51 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users
is there recomended source files for t.38 pass? latest cvs does not work
for me.
is it possible
publish working src?
turby
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Title: Snom 320 and message retrieve key
use
exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})
or universal
exten =
default,1,VoicemailMain(${CALLERIDNUM})exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})exten =
unknown,1,VoicemailMain(${CALLERIDNUM})exten =
hi hugh,
use script and run the script from rc.
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: 14. ledna 2006 1:51
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] loading zaptel drivers automatically upon reboot
is it possible only monitoring call between phone A and B from phone C?
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is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr ser91623 cdr)
exten = _00.,3,MYSQL(Query resultid ${connid} select\
I use 1.0.9 and 1.0.10
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
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Is it possible change in dialplan SRC value in CDR record? When I
change CALLERIDNUM, SRC field contain old value.
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ok, but I need this for 1.0.9
Do
Set(CDR(src)=value)
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gate:/etc/asterisk/.sys# cat astdog.sh
#!/bin/sh
#
#
sleep 60
#
while [ 1 ] ; do
BEZI=`ps auxx|egrep 'asterisk -p'|egrep -v 'grep'|wc -l`;
if [ $BEZI = 0 ]; then `killall -9 mpg123`; `asterisk -p`; fi
sleep 10
done
gate:/etc/asterisk/.sys#
---
turby
Is there anything I can set or any
is any problem with faxing trought:
PSTN FAX = PRI = ASTERISK = SIP/G711 = SIP ADAPTER
(like Linksys PAP2 etc.)
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Alejandro Vargas wrote:
btw. does anyone have a definitive list of all the finarea VOIP
companies? I can think of:
call1899
call18866
voipbuster
sipdiscount
voipcheap (note: this one uses a proprietary protocol, similar to IAX
but over
sorry, this is mistake
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