Displaying 20 results from an estimated 300 matches similar to: "IAX to IAX call with really bad echo"
2007 Nov 20
0
iaxpeers from Realtime
Hello asterisk users, here is a little problem pulling out iax peers from
real time database
I have the following peer configured in my database
mysql> select
name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit,
ipaddr,port from iax_users where name='iaxtermination';
+----------------+----------+----------------------------------+------+-----
2005 May 13
0
Problem with IAX trunking
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be not matching the accept packet with the relevant item in the iaxs
array due to the following
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
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
2008 Jul 25
1
v3.2.0 and MS-DOS: Directories must be upper-case??
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I have an MS-DOS client (Win98SE boot disk) connecting to a Samba v3.2.0
server which is a member of an AD domain (WinNT 5.0/2000 DCs). If it
tries to create a file in a directory, it will fail if the path is not
all-upper-case. For instance: J:\Bk-Full\DJohnson\S1565\ will fail but
J:\BK-FULL\DJOHNSON\S1565\ will succeed.
When it 'fails'
2004 Jul 06
3
Zap Channel error using 4-port FXO TDM400P
I have been having some troubles with the zaptel channel on what appears to
be the inbound process. The box is running the stable CVS code and has a
TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the
only active port on the card at the moment as we only have one analog line.
What has been happening is that it looks like Asterisk has been detecting an
inbound call even though
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", "")
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 Jan 13
0
Xfering a call
> Well that didn't work....I now get this error
>
>
> Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to
> create
> channel of type 'SIP'
> == Everyone is busy/congested at this time
> -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in
> new
> stackJan 12 16:56:21 WARNING[4989]:
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",
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),
2006 May 22
1
SIP to IAX - forcing codec pass thru
hi
We take calls inbound via SIP from our Cisco PSTN gateways, and pass it
to customers using IAX (they run their own asterisk servers).
We've noticed that asterisk is transcoding the call into a different
codec, if the customer prefers a codec different to that which our cisco
gw prefers. As such, the quality of the call can degrade.
We'd rather asterisk just passed through the RTP
2014 Jul 11
2
CentOS 7 Anaconda GUI resolution
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Greetings! I tried installing CentOS v7 (1406) to an old spare
machine, and the video card apparently don't play well with X. It is
incorrectly saying that either monitor I connect is only able to
support 640x480, when one is 1024x768 and the other supports a higher
(but now forgotten) resolution. In no way do I expect a fix to be
added just to
2004 Apr 07
1
Out of trunk data space on call number 16386, dropping
Hi all,
We keep getting these and all the calls between these two asterisk boxes get
dropped. what is going on here, I have been trying to solve this problem on
my own but maybe I don't have the trunk setup right. also I have posed the
output of my full log of the machine with the zap interface, the other is
using ztdummy.
IAX.conf on machine 1:
[general]
port=5036
;iaxcompat=yes
2004 Jul 05
1
FireFly client and echo problems with IAX
Hello,
I am having horrible echo problems when using the FireFly client on both the
caller and callee sides of the call. When I use another IAX soft client
like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone
else experienced this and do you know what might be the problem?
Thanks,
dj
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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
2003 Dec 09
2
Need help with jitter buffer/quality settings
I'm using Asterisk to do audio as well as H.263 video over SIP.
Actually the video works pretty well but I have trouble with the audio.
I'm wondering if someone can suggest codec/jitter settings or other
tweaks. The system looks like this:
Softphone <---ulaw----> Asterisk #1 <------IAX (usually GSM)--------->
Asterisk #2 <-------- IAX (usually GSM) -------->
2004 Sep 10
3
call quality monitoring
I need to debug a call quality issue with remote users on the other
end of a satellite link. The symptoms are: we here on the Internet
side can hear them just fine. On their end, things work sorta OK most
times, but they often suffer from severe dropouts and digital
warbling, both of which I attribute to them missing packets. Often
times they can't make out a word we are saying while we can
2004 May 23
1
IAX2 REACHABLE/UNREACHABLE
All,
I have an issue with IAX that I can't comprehend. Approximately every eight
minutes my servers go unreachable. They stay unreachable for exactly 10ms.
I have two servers running IAX and it happens on both servers
simultaneously. I have searched the archives and see similar issues, but
not the exact same one. I am on the current CVS stable version of *.
Also, during IAX calls,
2004 May 18
0
Asterisk to IAXTel help
I'm trying to make a call from an IAXPhone client - through the * PBX
to an 888 number using the IAXTel link. I'm using the basic conf files
for extensions and iax. I get successfully connected (at the
"Attempting native bridge" line of the output) and am then able to talk
both ways for 20 to 30 seconds and then the IAX phone appears off line.
If I wait on the PSTN line for