Displaying 20 results from an estimated 200 matches similar to: "Xfering a call"
2005 Feb 21
0
FWD problem
Guys.
Im using IAX and FWD and I think everything is setup fine.. someobdy just
tried calling me but my phone jus ran once and sent them straight to the
voicemail.. the logs show this:
-- Accepting AUTHENTICATED call from 65.39.205.121:
> requested format = ulaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (alaw|ulaw|ilbc|gsm),
2005 May 22
0
*@home 1.0 FWD inbound problems, 2 calls generated
Hi ALL
Have installed asterisk@home 1.0
On FWD DID's, appears that 2 calls are generated to the inbound extention. I
have confirmed this on a number of friends boxes also. Does anyone have a fix
for this ? I set the DID simply to a custom context and it did the same...
Anyone have a way to fix this ?
Here is the output......
-- Accepting AUTHENTICATED call from 65.39.205.121, requested
2005 Jul 10
0
iax fwd - calling twice
Hi,
testing a new fwd account, dialling from sip4030 to my FWD number,
sip4021 rings as defined in extensions conf.
Why is this happening twice?
-- Executing SetCallerID("SIP/4030-a7f2", ""HTCAS"") in new stack
-- Executing Dial("SIP/4030-a7f2",
"IAX2/617533:xxxxxx@iax2.fwdnet.net/617533|60|r") in new stack
-- Called
2005 Jan 12
2
Setting channel display in SIP
I have a situation where I need to know which Zap channel an incoming
call is on, so that the call can be answered appropriately when a SIP
phone displays the channel. These Zap calls are coming in over PSTN and
don't have caller ID.
As far as I can make out my SIP phones (WuChuan HOP-1002) display the
user part from the SIP "From:" header as the second line on the
display. If the
2005 Jan 29
2
Call rejected by FWD: Unable to negotiate codec
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the same settings works fine with *0.9)
I've standard settings in iax.conf
[general]
bindport=4569
register => xxxxx:xxxxxx@iax2.fwdnet.net
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
disallow=all
allow=ulaw
--
#Joseph
2004 Jun 26
1
IAX & FWD, No authority found?
Hi Folks,
Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded:
$IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036
I can make outgoing calls just fine, but when I receive an inbound call
2005 May 26
1
How do I diagnose the problem in this Asterisk test session with FWD?
=============
SJphone Log
============
Outgoing SIP session
Respondent: (sip:8612@192.168.2.2)
Remote client:
Started: May 26 16:33
Accepted: no
Ended: May 26 16:34
End reason: Call rejected: 503 Service Unavailable
===============
Asterisk Debug
================
Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r")
in new stack
--
2004 May 13
4
IAX Freeworld
I have looked all over the site(s) for help. But heres the problem. Im
missing something.
In coming works fine from FreeWorld via IAX. But when Dialing out i get:
May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I
don't know how to authenticate iaxtel to 65.39.205.121
my IAX.conf if as follows
[general]
port=5036
register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net
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
2006 Feb 23
1
not consistent log from asterisk
Hello,
I have 2 channels in iax.conf
[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes
[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xxxxxx
secret=***********
auth=md5
The problem is:
When I tell FWD to call me I have this output in my asterisk
consol:
Executing Dial("IAX2/iaxfwd-outbound-3",
2004 Dec 13
1
incoming call from pstn to fxo not working with Asterisk
When somebody call me on my pstn # cable connected to my fxo card it does
not work when I check my computer the following error shows
Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on
asterisk1 (pid = 2160)
Verbosity is atleast 3
-- Remote UNIX connection
-- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at incoming,s,1 failed so falling
2004 Aug 28
4
incomming call rejected using IAX2 with FWD
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?
Thanks,
S.
2005 May 07
4
Setting variable for a context for all extensions?
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it. I want something like this in
extensions.conf:
[from-iaxfwd]
exten => .,1,RING=r3
exten => 123456,1,Goto(from-pstn,s,1)
[from-internal]
exten => .,1,RING=r2
include => ext-local
[ext-local]
exten => 1,1,Dial(Zap/1,${LONGTIMEOUT})
exten =>
2005 Jan 28
3
FWD and IAX2
Hi,
I had a FWD account set up with asterisk (using SIP) and it was working
fine both ways. I switched to IAX2 and now I can't get incoming calls
from FWD. People who call my FWD number get a "480 - user is not online"
message without any traffic reaching my box. I can call FWD numbers fine
over IAX2.
It seems fwd isn't trying to place the call over IAX2 because it thinks
2004 Jun 21
1
IAXTel Help
I've searched WIKI and Archives but nothing.
Im getting:
-- Called username:password@iaxtel.com/1800somenumber@iaxtel
Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call
rejected by 69.73.19.178: Unable to negotiate codec
-- Hungup 'IAX2[Iaxtel]/8'
== No one is available to answer at this time
-- Executing Hangup("SIP/104-b8eb", "")
2004 Jul 02
1
IAX to IAX call with really bad echo
All,
I have spent the last couple of days looking through the mail archives and
the documentation on the Wiki, but have not been able to find a solution to
the problem. The version of code I am running is from CVS as of 6/30/04.
What happens is that when I make an IAX call to another IAX client the
caller receives a really bad echo. All of the documentation I found around
using
2003 Aug 07
1
[patch] New RC to differentiate partial xfers from files that get deleted before they're xfered
All,
During development of a backup solution with rsync I experienced some
failed backups because of RC 23, partial transfer. These were because the
application using the data I was backing up, was still active, and had
deleted a file inbetween rsync compiling the file list and then transfering
the file.
After some feedback from Wayne Davidson and JW Schultz suggesting the
solution to this,
2006 Mar 10
1
IAX / Firefly handshake problem
I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX
phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the
PBX is working fine, but the IAX phone still won't connect. Below is my
iax.conf and the output from setting iax2 debug while the phone tries to
connect. Could somebody please give me some pointers? This doesn't seem to
be a normal
2004 Sep 15
4
IAX to IAX connect question
Hi,
I got my * working fine with FWD at office with 2 extensions, i receive
calls and i can make calls thru FWD. I got also my * at home, and i
connected it using auth=rsa. From my home, i can make calls using my office
iax, but if i try to redirect incomming calls from FWD to my * at home, it
rejects the call. I created the pub/key pairs for rsa and its working ok
and i just pasted the
2009 Mar 19
0
Can I tell if a call picked up on PSTN extension... for example?
Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-)
My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL).
Calls come in and are