Displaying 6 results from an estimated 6 matches for "freephoni".
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2007 Apr 18
2
incoming SIP call
...t; asterisk SIP <=> lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport
From: "asterisk" <sip:asterisk@82.XXX.XXX.XXX>;tag=as01265eaf
To: <sip:freephonie.net>
Contact: <sip:asterisk@82.XXX.XXX.XXX>
Call-ID: 7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX
CSeq: 102...
2009 Nov 15
2
Sip incoming call issue with Asterisk 1.6
...stand the SIP debug traces, my server receives the
request and reject it:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
<--- SIP read from UDP:212.27.52.5:5060 --->
INVITE sip:s at 192.168.4.2:5060;transport=udp SIP/2.0
Call-ID: 25151-WW-0eaf098b-2f615ac60 at freephonie.net
Contact: <sip:172.17.20.241:5062>
Content-Type: application/sdp
CSeq: 239836027 INVITE
From: "096160XXXX" <sip:096160XXXX at freephonie.net;user=phone>;tag=25151-GA-0eaf098c-32a97dc05
Max-Forwards: 28
Record-Route: <sip:C=on-88.165.134.117.5060;t=RDKIW at 212.27.52.5:5...
2006 Jun 29
0
Asterisk with Sipbroker calling / routing problem
...time and yesterday I decided to add
sipbroker to my config. It was pretty simple and it works for some
numbers (e.g. I can call *258-9123, UK date & time - which is on the
"phone numbers you can call" page -) but fails for some others.
For example I've got a friend who's at freephonie so to call him, I
would dial *759608xxxxxxxx (7596 being freephonie.net).
When I do that, I get the following error:
Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '<sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe'
And her...
2006 Nov 27
0
[VoIP Trunk] No such host
Hello
I'm trying to add a VoIP trunk to Asterisk, but I'm getting the following
warning in the log file if I leave "srvlookup=yes" in sip.conf (OK if I
comment it out):
--------------
Nov 27 16:40:22 NOTICE[29660] chan_sip.c: -- Registration for
'087077XXXX@freephonie.net' timed out, trying again (Attempt #47)
Nov 27 16:40:22 WARNING[29660] chan_sip.c: No such host: freephonie.net
Nov 27 16:40:22 WARNING[29660] chan_sip.c: Probably a DNS error for
registration to 087077XXXX@freephonie.net, trying REGISTER again (after 20
seconds)
--------------
Any idea...
2008 Jan 07
0
chan_mobile and W300i
...mobile search)
;port=5 ;headset profile
context=frommobile
adapter=intuix ; adapter to use
dtmfskip=60
group=1 ; this phone is in channel group 1
more sip.conf
[general]
defaultexpirey=1800
dtmfmode=auto
qualify=yes
context=default
srvlookup=yes
register => _________:***********@freephonie.net
disallow=all
allow=ulaw
allow=alaw ; Allow codecs in order of
allow=ilbc ; preference
allow=gsm
allow=speex
[sip-out]
type=peer
host=freephonie.net
username=XXXXXXXXXX
fromuser=XXXXXXXXX
secret=**********
nat=yes
[sip-in]
type=peer
context=fromsipproxy
host=freephonie.net
[1000]
typ...
2009 Oct 05
3
Questions about app_jack.c
...dummy sound card (allow me dial command). I do a call with a
JACK_HOOK from app_jack.so, sound is sent but no one is received.
My extensions.conf :
exten => _0.,1,Answer
exten =>
_0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu
t)))=on)
exten => _0.,n,Dial(SIP/freephonie-out/${EXTEN:1})
Asterisk command :
console dial 0xxxxxxxx
2) Jackd works well with anothers applications when I force them to use jack
as input/output. -> probably not a jack configuration problem.
3) If I kill jackd and I use chan_alsa.so with the real soundcard, it works.
-> probably no...