Displaying 20 results from an estimated 700 matches similar to: "MWI with IAX"
2004 Dec 21
0
IAX2 insists on not using port 4569??
For some reason, starting just today, 1 out 3 of my asterisk servers is
having issues calling 1 other server. The only issue I see is that when
it registers with the problem server it is using port 1027, not 4569.
ie:
Registered to 'Server 1', who sees us as 'Server 2':1027
Server 1 then proceeds to timeout trying to register with Server 2.
The way I have each server registering
2007 Mar 02
1
Double DTMF digits sent on IAX native bridge
Hi,
I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users. The two servers connect with each other
using IAX. When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
channel bridge together. Now every time I dial a DTMF digit, the
asterisk is sending two DTMF digits. I enable
2006 Nov 22
0
iax2 - wildiax phone & myself puzzled
I know in advance maybe I'm overlooking something stupid,
however I'm really lost and cannot find the solution...
Situation:
- asterisk-1.2.13 on a linux box with no iptables active.
- one iax2 peer defined
- one wildiax phone running on my laptop
the soft phone is configured to connect & register on asterisk,
however, I cannot get it working.
What am I missing? Please help!!
2004 Mar 26
1
DIAX Followup
Anyway, in my P.S. yesterday (the main post was on Codec problems), I
described a situation where any IAX softphone was registering
successfully, and then having zero sounds heard on either side of the
call. Here is an "iax2 debug" output from a DIAX call to a local *
server, dialing the extension that goes directly to the "demo"
application.
AsteriskHouse*CLI> iax2
2007 Sep 14
2
AGI script fails on IAX channels (from call file).
Hi Guys,
I have already tried this one on the developers list. I have not been
successful getting much back there and I have notified them that I will
post this on the users list instead. Hopefully somebody have tried
something similar and can help out.
I am developing AGI scripts on Asterisk and have run into some very
strange behaviour and I think this is a bug, but I am not completely
sure.
2009 Oct 02
1
IAX2 Call rejected, CallToken Support required
Hi All,
I am using Asterisk 1.4.26.2 and I am getting the following problem
making connections to this server. My other servers are Version 1.2.x
which have no problems and this 1.4.26.2 server can call the other 1.2.x
servers.
The error is:
chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support
required. If unexpected, resolve by placing address 192.168.25.250 in
the
2006 Oct 18
0
IAX2 thru NAT problem
Hi people,
i have problem with IAX2 between two asterisk PBX. When i try call some
number i get "INVAL" packet, but when i try call same number via OpenVPN
(is between this two asterisk) call is working fine.So i debug
communications and here is my opinion ...
Schema of connection:
Asterisk1 -> ADSL router with NAT -> INTERNET -> Asterisk2
A)Calling directly via public
2007 Aug 04
2
IAX2 - DualServer Problem
Hi,
I have two asterisk servers and I want to make these servers call each
other as they were internal. I have succeeded in one way. Server B can
call Server A without problem, but Server A cannot call Server B.
Here's the iax configuration of servers
Server A:
==================
[ipek]
auth=rsa
context=from-internal
host=XXX.XXX.XXX.XXX
inkeys=ipek
outkey=odtu
peercontext=from-internal
2008 Mar 28
1
how to register IAX user without password
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid="Guest IAX User"
My extensions.conf
[default]
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
2006 Nov 01
1
IAX problem
Hi All,
I'm having problem with IAX, I'm trying to connect to speex.co.il from
asterisk using:
register => username:password@speex.dyndns.org
and I cant get it to work.
Maybe someone who already got this to work will help...
When dialing my speex extension I see the next output from consol:
IAX2 Debugging Enabled
*CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:
2004 Apr 01
0
I'm still a little lost...
I downloaded iaxComm and get up my iax.conf file and the
extensions.conf. Here is the out but from CLI in iax debug. What did I
forget to do???
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00001ms SCall: 10489 DCall: 00000 [192.168.50.66:4569]
USERNAME : 100
REFRESH : 300
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
2007 Apr 03
0
DTMF via IAX ignored after a few seconds
I'm new to this list, and I apologize if this is an already answered
question, but my Google-fu was not strong enough to find the answer if
it was.
I'm having a problem with DTMF on incoming IAX calls. For the first few
seconds of the call (between maybe 1 and 15, it varies from call to
call) everything works fine. After that I continue get DTMF_E messages
from the remote IAX server
2004 Oct 06
0
iax2, strange native bridge problem????
hallo,
i am really confused how nativ briging is working with asterisk,
i use a asterisk server as central server and register another asterisk and
an iaxcomm client to the server, all three have public ips on the internet.
somtimes, when i call from iaxcomm to my asterisk, the calls go peer to
peer (i can see it with tcpdump) but sometimes the get routed through the
central asterisk server
2005 Mar 05
0
Is anybody having problems with sixtel?
Hi,
My termination with sixtel stopped working, is it something I did or anybody
else is having the same problem.
I am attaching log:
*CLI>
-- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack
-- Goto (macro-dialout-default,s,4)
-- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack
-- Goto (macro-dialout-default,s,6)
-- Executing
2004 Apr 02
1
problems getting inbound to work @ voicepulse
Hello-
I'm obviously doing something wrong here in trying to get an inbound
DID to work with voicepulse.
I have an outbound context set-up for those calls in iax.conf, and the
appropriate register in- statement.
within extensions.conf I am doing something like this:
exten => 212xxxxxxx,1,Dial(SIP/admin,t)
(where admin is the phone i am looking to forward to from sip.conf).
i'm
2011 May 05
0
Could not place calls through IAX
Hello,
I have some problems in placing calls through IAX... It does not work :)
in the asterisk console I can't see nothing about dialplan enter or
so, IAX debbugging seems to be unuseful...
this is my configuration:
[612]
type=friend
secret=123456
notransfer=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=from-internal
host=dynamic
requirecalltoken=no
I enabled IAX debugging, but
2008 Mar 28
1
how to register IAX user without password for any user
Dear Sanjay,
Sorry sanjay i miss to explain completely. My PC2PC mean is
Dialer2Dialer i want to allow call between Dialer with out any
registry and authentication through IAX.
so i need to setup Asterisk accept calls from any user and users can
call to each other without any password and registration.
please help how can i configure Asterisk using IAX in this regards.
thanks,
Asif
Message: 9
2005 Feb 21
2
Conecting to asterisk server through NAT usingIAX
Hallo
Did you allow udp outgoing on 4569 as well.. i found
udp bit different than
tcp when comming to firewalls
liaan
----- Original Message -----
From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws>
To: <timebandit001@gmail.com>; "Asterisk Users Mailing
List - Non-Commercial
Discussion" <asterisk-users@lists.digium.com>
Sent: Monday, February 21, 2005 12:29
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
--