Displaying 20 results from an estimated 3000 matches similar to: "Call sip:user@domain.com?"
2010 Dec 13
1
Configuring server to call SIP numbers on the Net?
Hello
This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix... how should I configure Asterisk so
that SIP clients can dial SIP numbers on the Net, such as those below
to perform an echo test?
www.voip-info.org/wiki/view/Phone+Numbers
I tried pasting numbers in XLite, but nothing
2010 Dec 14
1
Asterisk + VOSP account working configuration?
Hello
I'm having a difficult time finding precisely what to put in
sip.conf and extensions.conf (and possibly other files) to get a
working configuration to connect an Asterisk (1.4) server to a VoIP
provider with the Asterisk server + SIP clients located in a private
LAN behind a NAT router:
http://img560.imageshack.us/img560/3749/asterisknat.png
Would someone have a full, direct (ie.
2010 Dec 29
2
Log and forward calls to cellphone?
Hello
I don't have a landine and use a VOSP to provide access to the
telephone network.
In case a call comes in and I'm not home, I'd like Asterisk to log the
call, and then send an SIP message to my VOSP so the call is forwarded
to my cellphone and is thus charged to the caller, without Asterisk
having to dial out to my cellphone through my VOSP at my expense and
bridge the two
2006 Oct 11
10
GPL Softphones
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network. I found Kiax but only
for IAX protocol.
Did you know a good GPLed softphones which works on Windows ?
Thanks
Greg
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2011 Feb 03
1
[newbie] Conference call
Hello
I've never used Asterisk for a three-person call, and would like to
check that MeetMe is the way to do this.
The ADSL modem provided by my ISP offers free calls to
landlines/cellphones when using a handset connected to an RJ11 port on
the modem.
A three-person call can be set up by using the standard PBX sequence:
1. Using the handset, call party #1
2. Hit "R" key on
2007 Jul 10
2
video calls - Windows / Linux interoperability ?
I will install Asterisk on my home server, I want to be able to route
video calls, but I need the Windows and Linux clients to be interoperable.
On Linux, it looks like Ekiga is a good candidate. But how about Windows?
Anyone using Kapanga in an Asterisk network that includes Ekiga? Are
these two interoperable?
I'm not necessarily looking for open source software, free for personal
use is
2004 Jul 06
3
H323 channel
Hello everybody,
my * box is connected to gnugk with H323 channel. If I call from an H323
EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
start but noisy (scratch) , then became ok for callee (SIP EP) but still
scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
No
2010 Dec 22
4
Asterisk hangs up call after 20s
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT router:
http://img252.imageshack.us/img252/3749/asterisknat.png
I'm having the following issue: When the _local_ XLite calls out the
remote XLite, everything works fine;
2010 Dec 13
1
Application to test STUN + broadband?
Hello
I was wondering if someone knew of an application that could check
that the user has a firewall and a broadband connection that will work
OK with Asterisk and VoIP.
The app would first perform some bandwith + jitter tests, and will
then call a STUN server to check that the firewall isn't symetric.
BTW, is Asterisk now STUN-capable, or is it still to map ports
manually on the firewall
2009 Aug 17
2
Accessing to ekiga.net through Asterisk
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.
/etc/asterisk/sip.conf:
; Outgoing to ekiga.net
[ekiga]
type=friend
username=MyUser
secret=MyPass
host=ekiga.net
canreinvite=no
qualify=300
nat = yes
stunaddr =
2007 Dec 02
2
Asterisk install beta testing/config help
I have asterisk up and running on a fedora system but
having trouble accessing system via softphone (ekiga
and xlite). Im a newbie to asterisk and was looking
for some help walking through this. I imagine 10 - 15
mins would be all needed to make proper config changes
needed. Once I get this setup I'd be interested in
discussing customizations and scripts so any
freelancers or companies welcome
2005 Mar 03
2
Calling hangup in background
Hi everybody,
I'm running an IVR menu with different languages setted up by user when
they enter this menu. What I want is when they hangup, asterisk sets the
default language (aka en) back.
I'm wondering which extension is called after a hangup in a background
command?
BTW my IVR menu is in a goto context.
--
Daniel
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell
2010 Dec 21
0
Friend/user/peer in plain English?
Hello
I've done some googling, but still puzzled at my working
configuration.
Apparently, a "user" can only receive calls through Asterisk, a "peer"
can only make calls, and a "friend" can do both.
If that's correct, I don't understand why my VOSP requires the
following settings in sip.conf to let my Asterisk server make/receive
calls to/from the PSTN:
2011 Mar 17
1
[1.6.2.5] Asterisk can't find MOH file
Hello
I thought I had things set OK to have Asterisk play FR files for
prompts and MOH, but for some reason, it still can't find them:
============ ll /var/lib/asterisk/sounds/
drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/
drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/
Note: fr/ contains core + extra + moh as downloaded from here:
2011 Sep 18
1
[1.6.2.9] Echo even when using headset?
Hello
I just set up Asterisk 1.6.2.9 through packages on a test host
running Ubuntu 11.04, configured sip.conf/extensions.conf, and
launched EyeBeam 1.5.20 to run the echo test.
For some reason, even through I'm using a headset, there's a lot of
echo and after a few seconds, it sounds like it enters a very fast
loop before the echo stops somewhat. IOW, unusable sound.
Here's a
2011 Mar 15
4
[1.4] Asterisk doesn't hang up?
Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
========== extensions.conf
;Play MoH for a few seconds, hang up, and
;check ChanIsAvail() able to detect when line idle again
exten => 8888,1,Answer()
exten =>
2007 Sep 12
4
ASTERISK BOX behind a filewall
Hi All,
I want to put a ASTERISK BOX bend a Firewall. So I have given below rules.
iptables -A FORWARD -p udp -d 192.168.101.30 -m multiport --dports
3478,4569,5060 -m state --state NEW -j ACCEPT
iptables -A FORWARD -p udp -d 192.168.101.30 --dport 10000:20000 -m state
--state NEW -j ACCEPT
iptables -t nat -A PREROUTING -p udp -i eth0 -d 1.2.3.4 -m multiport
--dports 3478,4569,5060 -j DNAT
2006 Oct 23
1
Asterisk conferencing features
Hello!
I'm new in Asterisk and I hope that my trouble is very simple.
We're implementing a Education Project of a e-Learning system (LMS)
that uses conferencing (video and audio) over internet.
The e-Learning system will be on GPL license, and for that, we're
using only free software to implement.
Asterisk is our first choice for video and audio conferencing, and
making tests,
2007 Sep 04
1
SIPBroker vs SIPgate
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is "we don't support SIPBroker"...
So whats the easiest way to support SIP <> SIP network calling?
At the moment, I've setup some local shortcodes (eg dial **777. to goto
sipgate.co.uk) based on what Gradwell