Displaying 20 results from an estimated 1400 matches similar to: "FWD IAX Problem"
2005 Jan 13
2
Firefly repeats registering to * server
This may not strictly be an asterisk question, but not sure where else
to post ...
I have an Asterisk test server setup with two firefly clients, one on
the local lan and one on an external ip address. Both clients are setup
the same way and voice calls work fine. The asterisk console reports a
"Registered" message for the external client at about one minute
intervals but the
2005 Jan 23
6
Autio cut off at beginning of call
I posted this question a while back, and I'm posting again in hopes that
someone has some ideas. Sorry if you've already seen this.
When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
VoicePulse Connect) I often find that after the call is answered the first
few seconds of audio are cut off (i.e. I don't hear the called party). This
usually results in the called
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
2003 Nov 18
1
DIAX - Can place a call, but can't be called?!
Greetings,
DIAX seems to work well placing calls, but I can't actually receive a
call . Here, DIAX (x305) "registers", then I use a sip phone to place a
call to DIAX (which definitely is not in use by me at debug time, but it
is idle on my desktop.I think), and then * goes to vmail.
Here's the debug output:
affinity*CLI> iax debug
IAX Debugging Enabled
Rx-Frame Retry[N/A]
2004 Aug 10
1
Firefly and *... Argh!
Okay, I've read as much as I can, and I think i've followed
instructions, but I'm still having problems with * and firefly... I can
get outgoing to other freshtel working, but not incoming (I get the "not
available" voicemail), or outgoing to landline.
I'm using the debian asterisk package (0.9.1-RC1-4)
My iax.conf has in general (under my FWD register, which
2015 May 13
1
registering IAX with Teliax
Hopefully this is really a generic question about IAX and doesn't turn out
to be something specific to Teliax, because they haven't been too helpful
so far. All they can tell me is that my login shows "status unknown" on
their end, which prevents me from receiving inbound calls on my Teliax
number. Outbound calls through the same server work fine, which rules out
most networking
2005 Mar 06
1
IAX - Registration Problems
Hi everyone,
THis is my second thread regarding the issue.(before I was having problems
with accessing my email, which slow down my responses, sorry for that)
My setup looks like this
Firewall
|
|
Asterisk---Asterisk (two asterisk servers with the same setup for high
avail)
|
|
phones
Ports 5060, 10000-20000, 4569, 5036 are forwared to 192.168.1.251 which is
virtual ip address on one of the
2004 Apr 02
2
All calls go to Voice mail and never ring.
I'm starting to get this to work! Well I got Voice Mail to work!
All calls goes to voice mail without ringing the users phone (iaxComm).
Here is my iax.conf and my extensions.conf
Any help would be great!!
Thanks
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2009 Jun 01
1
Asterisk 1.4.26-rc1 Now Available
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release is primarily a fix for an issue (#14867, #14717) related to
security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid
registrations. Instead it sent
2009 Jun 01
1
Asterisk 1.4.26-rc1 Now Available
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release is primarily a fix for an issue (#14867, #14717) related to
security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid
registrations. Instead it sent
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:
2005 May 16
1
ShoreTel 210 MGCP phone drops calls with MGCP RSIP
I've got a ShoreTel 210 MGCP phone drops calls. My packet
capture indicates that the phone may be trying to renew its registration
with *, but reports Restart Method of Disconnected (frame 2), then *
seems to take that as a sign that it has lost the connection and closes
things down. The phone, meanwhile, seems to think it can continue the
conversation until a few ICMP "port
2009 Aug 27
1
Documentation on RSA key authentication ?? (No way to send secret to peer)
Is there any documentation on IAX RSA authentication because I followed
http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and
it's not working...
Asterisk 1 :
-r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub
-r-------- 1 root root 963 Aug 24 19:38 server1.key
Asterisk 2 :
-r-------- 1 root root 963 Aug 24 19:53 server2.key
-r--r--r-- 1 root root 272 Aug 25 09:02
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
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 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:
2005 Feb 03
1
MWI with IAX
Does the MWI feature work with IAX2? I have read where it should but
cannot get the indicator to work on any of the IAX softphones that I
have tried which have this feature. I even did an IAX debug and did not
see where and indication was sent to the phone when it registered.
IAX2 registration session:
*CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: REGREQ
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 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
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