Displaying 20 results from an estimated 1000 matches similar to: "Can't seem to connect/call fwd network Help!"
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 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",
2004 Jul 06
4
Newbie's doubt on sip.conf
Hi,
I have some doubts on sip.conf.
1) Can I have two or more SIP phones acting as
extensions in one Asterisk box, and at the same time,
registered to a SIP proxy, say Free World Dialup? If
yes, how?
2) Why we need a section in the sip.conf for the
proxy, say, Free World Dialup's fwd.pulver.com? In the
case of 1), how to assign the value to section
[fwd.pulver.com], since there are more
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
2004 Sep 16
1
Unable to dial using SIP using FWD and iConnectHere
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
----------------------------------
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to
get IP address for
2007 Jun 25
0
four ringing and hangup with error
Dear All
I have this setup
[asterisk]----[mediant2000]-------E1 Trunk----------[Avaya]
When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error
*CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip
2007 May 06
2
Were i make mistake
I've found some manuals and tried this to do :
Sip.conf
[test]
type=friend
username=test1
secret=test1
host=192.168.1.238
context=tutorial
fromuser=SIP Phone
callerid=101
nat=no
canreinvite=yes
dtfmode=info
disallow=all
allow=ulaw
[test]
type=friend
username=test
secret=test
host=192.168.1.240
context=tutorial
callerid=100
nat=no
canreinvite=yes
dtfmode=info
2004 Jun 01
1
SIP vs. SIP :-(
I'v a sip client and a sip trunk to FWD:
[general]
port=5060
context=default
tos=reliability
disallow=all
allow=ulaw
careinvite=no
[freeworlddialup]
context=default
type=friend
username=MYUSERNAME
secret=MYPASSWORD
host=fwd.pulver.com
[igor]
type=friend
callerid="Me"
host=dynamic
dtmfmode=rfc2833
careinvite=no
When i try to call a FWD number from SIP client i obtain a lot of
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
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 =>
2007 Jul 20
2
Announcing Digium/Asterisk World's Conference Program
Is this replacing Astricon this year?
If so it looks like a pretty poor showing in comparison to Astricon
Dallas last year.
Cheers,
Dean
________________________________
From: Carl Ford [mailto:carlf at vonmag.com]
Sent: Wednesday, 18 July 2007 9:09 AM
To: Dean Collins
Subject: Announcing Digium/Asterisk World's Conference Program
2003 Aug 12
0
RE: Asterisk-Users digest, Vol 1 #1033 - 7 msgs
Same thing. It will make sense to try
Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082
but in that case Asterisk sends
REGISTER sip:fwdnat.pulver.com SIP/2.0
which is not right. It should be sip:fwd.pulver.com but sent thru
fwdnat.pulver.com:5082
BR Borut
-----Original Message-----
Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT
From:
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 Jun 02
1
DTMF and SIP
Hi
I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the
phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also
tried inband) and I get the following error:
june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
This means that I cannot get access to voicemail from the handsets
2004 Dec 02
0
Connection Problem
Hi,
My configuration: Sipura 2000
Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux
I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never
2003 May 31
0
register with outbound proxy from behind nat for freeworlddialup etc.
Hi,
I've posted a simular message little over a week ago so sorry for
reposting. I need to register to freeworld dial up from behind a nat.
Using the xten software sip client works fine but with asterisk I don't
know how to do it. Last time I posted I got different responses. Some
saying I can't register with an outbound proxy from asterisk others said
they have done it. If it is
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 Sep 28
1
Forwarding SIP over IAX problem: No One Available
I'm hoping someone can help me out with this. I am basically just
trying to separate my outbound and inbound calls into separate contexts
instead of having everything in a single context. Any help would be
appreciated. Perhaps I've missed something really obvious....
Here is the network layout:
<remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>
2007 Aug 06
0
Digium|Asterisk World
Too bad it is August 6th
*P.S. Remember, as a member of the Digium Family we have secured a
special discount of 50% off of the conference fee for you if you
register by July 29, 2007. To take advantage of this limited time offer,
please register here
<https://secure.pulver.com/cgi-bin/von?mode=gpur&conf=dawfal07&type=g&pricode=billm>!*
Digium, Inc wrote:
>
> If you