Displaying 18 results from an estimated 18 matches for "pasqu".
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pasqua
2006 Oct 18
0
[OT] Nokia E60/61/70 and SIP
....conf is:
############################################
[208]
username=208
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes (I have try with neven too)
notransfer=yes
mailbox=208@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Pasqu <208>
disallow=all
allow=alaw
##########################################
I have set my phone with this conf:
##########################################
Public user name: sip:[PhoneNumber]@:[youre asterisks servers ip] or
sip:[PhoneNumber]@[youre_asterisk_servers_hostname]
Use Compressio...
2006 Feb 27
3
Asterisk with HT 488 FXO
...4
To: <sip:400@192.168.1.157>;tag=3a7300003fa70000
Call-ID: 5308aeca055e74a23ebe819d52d03d26@127.0.0.1
CSeq: 120 REGISTER
User-Agent: Grandstream HT488 1.0.2.16
Content-Length: 0
-------------------------------------------------------
The register string ??
Can anyone help me??
Thanks
--
Pasqualotto Enrico
email: pasqu@linux.it
web: http://www.pasqualotto.org
-----BEGIN GEEK CODE BLOCK-----
Version: 3.12
GIT d? s: a-- C+++ UL++++ P L++ E--- W++ N++ o K- w---
O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+
G e h++ r+ y+++++
------END GEEK CODE BLOCK------
2006 Nov 22
1
qualify=yes
hi all, how can I set the interval in second from retrasmit the magic
packets when qualify is set to on?
I want to view whitch voip-phone is connected but I don't want to DOS my
adsl connection.... ;)
Thanks Enrico P.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
2006 Jun 14
0
Asterisk & wengophone
...ssible to configure other sip
server (es. Asterisk) but every login wengo download from his site the conf.
Now I want that wengo download the conf from my http server with my
conf. :)
Now I work on this using patient and ethereal, is anyone make wengo and
Asterisk work or make this test?
--
Pasqualotto Enrico
email: pasqu AT linux.it || enrico AT pasqualotto.org
web: http://www.pasqualotto.org
skype: epasqualotto
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2007 Jan 10
1
Asterisk HA
...erience for this type of solution?
Is better ultramonkey, dundi or SER proxy in front of * server?
Thanks Enrico
P.S. Now during all this year I have to work with this type of solution,
why not make a fork of this ml .... for example
asterisk-ha@lists.digium.com, for write some docs too.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
2007 Jan 28
0
PHP sip client
...But I want to add a features....
I want to hang up the cornet of my phone, compose the number in my
browser and call a second phone.
In witch way I do this? Can i do this?
quickly... I want to replicate the numeric keyboard of my hw phone!
Thanks in advantage and sorry for my english. :(
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
2007 Feb 05
0
*****SPAMZ***** Asterisk cluster - keep up connection?
Spam detection software, running on the system "placebo", has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
enrico@pasqualotto.org for details.
Content preview: Hi all, how can I set up an asterisk cluster (using SER
or hearbeat) that keep up call and conference room when one of * server
go in fail state? There is a way that I install my asterisk software on
a virtual machine compose of N * server? [...]
Co...
2007 Feb 09
0
*****SPAMZ***** Conference & Page question
Spam detection software, running on the system "placebo", has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
enrico@pasqualotto.org for details.
Content preview: Hi. I'm currently setting up a particular conference: 3
members (a,b,c), a can speak with b and c, b and c can speak only with a
and not between them. I found my possible solution with paging/intercom
using option "d" (full-duplex), but...
2007 Feb 21
0
IAX Realtime - show peers works?
...ddr`, `port`, `regseconds`) VALUES
('ctm2', 'trixbox', 'friend', 'XXXXX', NULL, NULL, NULL, NULL, NULL,
NULL, NULL, NULL, NULL, 'fromiax', NULL, '10.0.0.254', NULL, NULL, NULL,
NULL, 'yes', 'all', 'alaw', NULL, 4569, 0);
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
2007 May 08
0
Beronet card - issue?
...o=no
nodialtone=no
callgroup=1
pickupgroup=1
presentation=-1
screen=-1
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=from-pstn
msns=*
This is the first time that I configure this type of card.... Link of
some good docs is ok too. :)
Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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2007 Jul 16
0
Dial and option G
...rnal]
exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1)
[fromiax]
exten => 9999,1,MeetMe(9999,qdxAa)
exten => 9999,2,MeetMe(9999,qdx)
exten => 9999,3,Dial(other-user,,G(from-iax,9999,4))
exten => 9999,4,MeetMe(9999,qdx)
.... but not work.
Any suggestion?
Thanks Enrico
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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2007 Oct 01
0
Park problem on IAX2 channel
...able to find extension '77' in context 'from-internal'
appears also with '#', '#7', '', '0'...
It seems that the dtmf came across the iax channel and arrive to other
asterisk.
The are a way to block this dtmf across the IAX trunk?
Thanks Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto
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2007 Oct 05
0
Asterisk translator issue?
...his time (1:0/0/1)
instead I use only alaw.
Infact if I keep che codec_ulaw.so and during a call watch the used
codec all are alaw.
Anyone can explain me where is the problem?
P.S. for me is not a problem to keet one file but is interesting to know
who want to translate who.
Thanks Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto
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2007 Oct 29
1
Realtime & context
...atabase the context also? Now if I want to
add a new context I have to modify the extensions.conf with:
[newcontext]
switch => Realtime/@
but I have about 50 asterisk that read one database, now if I want to
change/add a context I have to change 50 extensions.conf file :(
Thanks Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto
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2008 Jan 22
0
Conference Hangup
...i all, I have a question on asterisk conference.
Now I use appl Meetme with option A & x because when a marked person
hangup I want to close all the conference.
But what I have to do if I want two marked person and kill the
conference when one of two hangup?
Is possible?
Thanks. Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto
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2008 Oct 16
0
app transfer problem
...cenario is 3 asterisk box connected with iax trunk, I talk from BOX1
to BOX2 and I want to transfer user on BOX2 to another user on BOX3.
After transfer command is executed on BOX1 (return SUCCESS) on other BOX
I don't see nothing. Only hangup.
There are some way to debug the transfer?
Thanks Pasqu.
2007 Feb 09
0
Conference & Page question
...=> Debug,1,Noop(dialstr is
LOCAL/PAGE4441@ext-paging&LOCAL/PAGE4442@ext-paging)
exten => 4446,1,Set(_FORCE_PAGE=1)
exten => 4446,n,Macro(user-callerid,)
exten => 4446,n,Set(TIMEOUT(absolute)=60)
exten => 4446,n,Page(LOCAL/PAGE4441@ext-paging&LOCAL/PAGE4442@ext-paging)
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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2007 Feb 14
0
Asterisk & CME integration using h323
...192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhos== H.323 Connection deleted.
I don't understand why the call goes down only from cisco to
asterisk.... any ideas?
Thanks Enrico
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto