Displaying 20 results from an estimated 1100 matches similar to: "here's my IAX callthrough app and some questions about problems I have."
2005 Aug 04
1
Getting asterisk to work with callthroughs?
Hi,
Firstly, what I'm trying to do is:
* Get asterisk to pick up a SIP call via a DID
* Prompt the user
* When the user puts in a number, go to IAX.conf and route it according to
what I've specified there, i.e Least Cost Routing, etc.
I've set-up something similar to what I've found online, but it doesn't
work! Asterisk doesn't pick up the call at all..... :(
The files
2016 Jun 30
2
problem with DTMF detection on calls created with Originate AMI command
Dear all
i'm creating an outgoing call to number xxx with this command:
http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external
&Exten=testDTMF&Context=cRETEUNICA&Priority=1
wich points correctly to this portion of dialplan:
[cRETEUNICA]
exten => testDTMF,1,Answer
exten => testDTMF,n,Read(digito,,1)
exten => testDTMF,n,SayDigits(${digito})
The
2005 Feb 08
1
Asterisk causing server to hang ... any hints?
I am trying to set up a simple Asterisk server. All
it's going to do for now is to act as my voicemail
box. I've got a DID from Voicepulse, and am using IAX
(I'll get to SIP someday when I want to circumvent the
phone company for long-distance, but for now I'd be
happy to get a trial version of Asterisk running).
So far, I've managed to set up voicemail.conf,
extensions.conf
2004 Aug 30
2
VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
A caller to my DID's hears Asterisk, but pressing DTMF does nothing:
On call setup "iax2 debug" shows:
-----------------
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
2005 Feb 19
1
sending traffic to LiveVoip
I have several DIDs (working well) with LiveVoip and I just signed up for
some outbound minutes. Unfortunately they did not send connection
instructions.
I tried:
exten =>
_1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s)
but I get the error
Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected
by 217.160.244.186: No authority found
--
2013 Apr 08
3
extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
sip3.voipvoip.com:5060
2006 Oct 24
2
IAX2 goes "one way audio" when lag gets bad
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I have the following config in iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all
2005 Jul 02
1
Problem registering Asterisk Dual Server
Here is my configuration everything would seems be straight forward, but
I can not register both asterisk with each other.
Both asterisks have Static IP but they are behind firewall/router, so
it means I have to use Register statement.
I'm a bit confused about the register statement.
How can they can register with each other when both firewalls are
blocking port 4569?
Do I have to open
2005 Sep 08
10
voice over atlantic
Hi-
I'm using IAX between two boxes, where one box is located in US and the
second in Europe. I'm trying to achieve the best voice quality and
mainly reliability between these boxes and looking for hints and
experience of others.
Facts:
- Asterisk 1.0.7
- RTT varies from 130-170 ms, depends on time and actual Internet
throughput
Questions:
- What is the sugested codec for such setup?
2006 Oct 16
3
Why is this happening?
In my IAX config file I have:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1
notransfer=yes
allanrobertson- 209.23.224.97 (D) 255.255.255.255
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
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
2007 Apr 20
6
How can I improve call quality?
Hi All,
I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth). We've no E1/T1 links, everything is IP based.
My boss complains that many of the calls he holds with others has a bad
quality. He also says that its not just him.
My iax.conf file
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 Nov 03
3
Problems Overwriting CallerID with True ANI
I receive calls over a T1 with callerid and then *ani*dnis*. I am able
to strip out the ani and the dnis in the dialplan but when I try to set
the caller ID to be the ani, it looks ok but then if I do a NoOp
callerid on the next line, I get unknown.
Here is the section of my dialplan:
exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN})
exten =>
2007 Aug 21
1
SET EXTENSION
Hello All,
How can I SET EXTENSION from context?
This is my context: -
[docall-usa]
exten => _NXXNXXXXXX,1,Answer
exten => _NXXNXXXXXX,n,Set() ; <<What do I need to set here>>
exten => _NXXNXXXXXX,n,DeadAGI(dousacall.php|1)
exten => _NXXNXXXXXX,n,Hangup
I need to add 1 in front of ${EXTEN} and then send the call to dousa.php.
Set(CALLERID(number)=1${EXTEN}) will set
2005 Aug 19
4
Overriding Caller ID
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible
2006 Jan 06
3
transfer application
I am having trouble understanding how to use this. I want to transfer
certain incoming calls from an IAX ITSP based on caller ID. From what I
can make of the docs, I thought I need to do something like this...
exten => _NXXNXXXXXX,n(nocid),transfer(1000)
exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS})
exten => _NXXNXXXXXX,n,hangup
exten =>
2008 Oct 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
2010 May 05
2
Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with "unavailable". There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXXXXXX
exten =>