Displaying 20 results from an estimated 1000 matches similar to: "IAX & WaitExten"
2006 Nov 01
5
DTMF over IAX
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
2015 Feb 05
1
IAX2 problem for WAN connections
> On Thursday 05 Feb 2015, jg wrote:
>> Calling from ServerB to ServerA works, but not vice versa. The only odd
>> thing that appears to me is the different perceived port on ServerA.
>>
>> Does someone have an idea at what to look in detail?
>
> Look in /etc/asterisk/iax.conf in the first instance.
>
Basically I used the example from the Asterisk book
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
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 Nov 29
2
Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both DiDs!
Thanks,
H
My iax.conf is below. When I dial the DiD provided by ITSP_B, the
other
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
2005 Mar 25
2
WaitExten question
I'm a bit confused about how WaitExten works. I assumed that when it
returns 0, the next priority in the extension would be executed, but
that doesn't seem to be the case. When I get to WaitExten and enter
extension 8, it plays the message, then Waits another 10 seconds and
times out.
[local]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten =>
2009 Sep 02
2
DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Is there any known reason that the DISA() routine should behave
differently than WaitExten() as far as recognizing DTMF tones? If
not, I suspect there's a bug here.
Try it yourself--two DID's on our PRI, numbers below let you test each routine:
It is my observation that some setups/phones DO and some DO NOT
express this variance.
--I could not show any variance on a sprint mobile phone
2011 Mar 23
2
using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten =>
2013 Sep 06
1
11.4.0: iax packets lost by amazon ec2
I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get
iax to work.
I've opened 4569 in the EC2 Security Group.
I'm using the zoiper client. Using tcpdump I can see the zoiper packets
coming in on 4569, but nothing shows on the asterisk cli.
Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on
interface 0
0000 12 31 3b 12 40 84 fe ff ff ff
2009 Nov 15
1
Call IAX2 => "Call rejected, CallToken Support required"
Hi
i have a small problems on two Asterisk Server 1.6.4 :
The first sent the call to the second, and in the second, i have a error :
[Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call
rejected, CallToken Support required. If unexpected, resolve by placing
address IP_FIRST_ASTERISK in the calltokenignore list or setting user
04TELNUMBER requirecalltoken=no
on the second
2007 May 16
1
WaitExten not responding on key presses
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten => 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten => _X.,1,Set(CALLERID(name)=Hotline)
exten => _X.,n,Set(original_extension=${EXTEN})
exten => _X.,n,GotoIf($[${announce}=1]?4:10)
exten =>
2008 Feb 02
1
Echo() app doesn't work
Hello list,
New to asterisk and to the list (although experienced in Unix/Linux
administration).
Short problem description:
--------------------------
I cannot get the Echo() application to run on any 32bit platform I can get my
hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have
runs just fine. In all cases asterisk log shows the same -- that Echo() is
executed
Details:
2006 Aug 28
3
lost packets when bridging zap and iax
We have a machine with a TE410P in it acting as a client to route calls
via iax2 to our central server,
caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called
client server
often the called can't hear the caller (both machines on public ip)
'iax2 show netstats" on client machine shows more and more dropped
packets on the
2007 Jul 27
3
Need help with inbound IAX
I have just started working with Asterisk and have run into a road block
concerning IAX and an inbound DID from callwithus.com. I am getting
nowhere and I don't really know how to isolate the problem. The asterisk
version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can
connect and make a call to other internal extensions using zoiper and
iax. When I try and use the number,
2006 Nov 19
2
WaitExten only reading 1 digit.
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
"Waitexten" app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make Waitexten wait for 3 digits?
I have setup the extension "100" for users to reach the
2008 Jan 09
3
WaitExten and Macros
I am trying to use a WaitExten in a Macro, and I am finding that the extension which is pressed ends up in context of the calling context and not in the Macro.
How do you do a WaitExten in a Macro?
Tony Plack
2004 Jun 18
3
WaitExten substitute
i am using the freebsd port, which seems to not yet have WaitExten(),
which i kinda want to use thusly
[ext-666]
exten => _.,1,SetVar(areacode=666)
exten => _.,2,Background(zz-in-who) ; give them list of extns
exten => _.,3,WaitExten(10) ; let them enter extn to call
include => extensions
include => applications
include => speeddials
2009 Jul 03
1
DTMF is not working occasionally over IAX Trunk
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digium card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients are connected to third server. A is the PSTN termination
server, B runs the menu and AGI and C is where
2005 Sep 22
1
WaitExten
Hi,
In my dialplan I'm using a WaitExten() command. It works only with Zap
phones. When I dial this command with Sip phone asterisk do nothing.
Should I put extra definition in sip.conf to make this work with Sip
phones?
Thanks in advance
Cheers