Displaying 20 results from an estimated 7000 matches similar to: "Newbie's doubt on sip.conf"
2004 Dec 18
4
Free World Dialup and Asterisk
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD
2003 Dec 24
2
FWD problems
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I had one before, but I lost
it.. :)
I've got it running through Asterisk - all working fine
2003 May 24
4
Free World Dialup behind NAT
Hi,
after reading about it on the list I decided to set up a Free World
Dialup account. For those of you who don't know, that is a sip proxy
where you and your friends can singn up free and then you can just
connect to it with any sip client and call anybody that is registered
for free. Pretty much like iaxtel (I belive that was the name of it) for
the iax protocol. It even supports clients
2008 Nov 20
2
Any other "free" toll free SIP providers out there?
FWD (Free World Dialup) allows any SIP call to US toll free numbers via *
18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at
FWD so in my dialplan I have something like:
exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r)
exten => _8.,2,Hangup
And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine
99% of the time.
I'm wondering if
2003 Dec 20
2
More beginner questions
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to
the internet port 5060 being forwarded to
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX
2003 Aug 19
1
Problem with * server and FWD
I have a small HUGE problem with *.
I have installed * but I have 2 problems.
1 - Making call to FWD.
2 - Receiving call from FWD
More info of the problem at the end.
Here is the sip.conf file.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ;default Default for incoming calls
register =>
2004 Jan 17
1
Registering multiple FWD accounts
Can multiple FWD accounts be registered?
I have the following output in my sip.conf file:
register=74928:xxx@fwd.pulver.com/74928
register=75160:xxx@fwd.pulver.com/75160
register=74573:xxx@fwd.pulver.com/74573
[fwd-74928]
type=friend
secret=xxx
username=74928
host=fwd.pulver.com
[fwd-75160]
type=friend
secret=xxx
username=75160
host=fwd.pulver.com
[fwd-74573]
type=friend
secret=xxx
2004 Sep 14
1
Setting up Asterisk with fwd
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial into it from
another fwd # it says user is not online.
In sip.conf I have the following added:
register => xxxxxx:xxxxxx@fwd.pulver.com/489125
[fwd.pulver.com]
type=friend
2003 Dec 12
1
simple question on sip.conf
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another SIP proxy can
redirect his calls to my * as well. Those calls two will come through
general
2004 Jul 04
2
I wanna kill FWD.... GRRR!!!
I'm a bit on the pissed off side right now.
If you call Free World Dialup conf bridge.. "514" is the FWD number.
Ok last night I was trying to talk to a few people in the conf bridge
because they msged me saying they were there. Ok I call it says "Please
Hold" then says "All circuits are busy"... Ok fair. I call like 25 times.
Well needless to say FWD side
2003 Jun 25
4
Asterisk and FWD
I can't get my Asterisk to register/place calls with FWD. Here's what I have
in my SIP.CONF:
register => 11111@fwd.pulver.com/11111
[fwd]
type=friend
secret=somesecret
host=fwd.pulver.com
username=11111
fromuser=11111
fromdomain=fwd.pulver.com
I'm using CVS version of Asterisk, checked it out last week. I get
authenticate error when registering with fwd, and all my calls to
2005 May 13
4
1-800 with FWD
Sirs,
Thank you for your quick response.
But when i try to make a call to FWD the following error appears:
For example, when i call to 612 (a service number of FWD)
-- Executing Dial("SIP/Phone4-e85b",
"SIP/612@fwd.pulver.com|90|Ttr") in new stack
-- Called 612@fwd.pulver.com
-- Got SIP response 500 "I'm terribly sorry, server error occured
(1/SL)"
2003 Aug 12
1
Using Asterisk with FWD through NAT
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
I tried with
Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com
but since Asterisk still use internal IP in some SIP fields I got "479 We
don't accept private IP contacts. Please set your external
2004 Aug 07
2
Asterisk : No Sound No Dial
Thanks for taking a look greg and hank. This seems to be getting bettre
everyday..help please
My sjphone is running on the same box as asterisk...i believe then the red
hat firewall should not be a problem.
Whenever i dial from CLI i get
#########
Executing Goto("OSS/dsp", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Wait("OSS/dsp",
2005 Aug 13
1
receiving calls from FWD
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming calls.
This is the relevant part of sip.conf
[general]
.....
register => 688426:xxxxxxxx@fwd.pulver.com/6000
[fwd.pulver.com]
type=friend
username=688426
fromuser=688426
2004 Jun 23
5
Really basic stuff :(
Hi :)
I've had all this working before, but I'm revisiting it, and in short, I
currently have huge problems receiving incoming calls. I've been trying
with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel
and libpri as of yesterday afternoon.
Would someone mind helping? :)
My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set
as the 'DMZ
2004 Oct 07
1
Confused about NAT and Authentication with FWD
I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT.
The problem:
I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem.
Now if in the extension.conf file I have,
exten =>
2006 Feb 07
1
asterisk to FWD
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
----------------
[internal]
exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD)
exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD
exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD
2003 Jul 10
1
Voicemail answers, but drops SIP call after about 3 seconds.
I am calling from my laptop to an asterisk box which answers the call
and I can hear the voicemail prompts, but the problem is that after so
many seconds, MSN Messenger drops the call because it thinks it hasn't
been answered by the remote machine. I'm not sure if this is an
asterisk problem, or if it is Messenger not knowing the call was
answered.
Has anyone else run into this sort of