Displaying 20 results from an estimated 600 matches similar to: "call file vs. originate"
2007 Dec 10
0
diferents events between ast1.2 & ast1.4 ??
Hi all,
I'm new in the list, and I have a problem upgrading from asterisk 1.2 to
asterisk 1.4:
There is a diference from asterisk1.2 to asterisk1.4 in AMI events.
When I do a call to a queue (with the same extensions.conf dial plan)
with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4
apper only 2.
It is normal? anyone knows it? what is the reason?
I
2009 Jun 30
0
Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE>
Originally posted on asterisk-dev with no response for 5 days, so posting it
to the wider audience now.
Asterisk Release 1.6.1.1
Scenario:-
1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and
902
2. Using AMI, 901 is Originated
3. When 901 answers, it is Redirected to an extension "exten =>
dial,1,Dial(SIP/902)"
4. 902 rings, then answers
5.
2005 Jan 21
0
Help DIALSTATUS gives ANSWER when line is BUSY?
I'm running Asterisk CVS-v1-0-12/20/04.
I'm using PHP with Manager API Here is the code:
####################################################################
# Make call
####################################################################
$socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout);
if (!$socket) {
echo "$errstr ($errno)<br /\n";
} else {
2011 Apr 08
1
Documentation for Asterisk AMI Events?
Hi Everyone,
I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is
specific to 1.6. I am wondering if the developers cared to write about the
new events that are spit out in Asterisk 1.8 somewhere on the web?
I checked the tar ball for asterisk 1.8 and documentation doesn't include
this event:
*Event: Unlink*
Privilege: call,all
Channel1: SIP/9999-00000029
Channel2:
2008 Feb 20
0
Strange NewCallerIDEvent after channel are linked
Hi all,
just for learning purposes i made a little gui frontend that visualizes
incoming and outgoing calls in realtime, using the events of asterisk.
I experienced a strange behaviour for outgoing calls. The callerid for
the *called* person got changed to one of my own numbers, after the
channels git linked.
After looking into the flow of events i saw that asterisk keeps sending
an
2006 Jun 13
1
calleridname.agi patch to only overwrite name if it is missing
I edited the calleridname.agi patch to only overwrite the name if it is missing.
The asteridex option still overwrites the name since it is our master list for known numbers.
--
Steven
calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006
@@ -16,6
2007 Aug 27
0
Bad hangup event cause
Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19.
Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command.
This is the 'sip debug' output:
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
INVITE sip:1
2005 May 25
2
Manager and Callerid problems
Guys.
Anybody knows why this is happening? Seems every time I make an internal
call, the manager shows this and I don't get the callerid on my identapop
but rather the calledid..
Event: Dial
Privilege: call,all
Source: SIP/intruder1-85f0
Destination: SIP/test-f037
CallerID: 201
CallerIDName: Anton Krall
SrcUniqueID: 1117038116.7
DestUniqueID: 1117038116.8
Event: Newchannel
Privilege:
2007 Oct 24
1
AMI ActionID.... Doesn't work
Is it well known that setting the ActionID when connecting to AMI has absolutely no effect?
Is this fixed in Asterisk 1.4?
If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much
2005 Mar 10
3
SetCallerID({$NEWCALLERID})
I am trying to SetCallerID to a variable I have defined. This obviously is
wrong. It actually sets the caller ID to $NEWCALLERID. I have search
through the examples on wiki but wasn't able to find something similar to
see what I was doing wrong. Could someone tell me the correct way to
SetCallerID to a defined variable?
exten => 2125551212,5,SetCallerID({$NEWCALLERID})
exten =>
2008 Jan 14
2
What is connect-debounce wrt usb?
I get the following message on a Centos 5 system (really a Trixbox 2.4
build on Centos 5):
Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port
1 disabled
What does this mean?
This message occurs about 30 times/sec for about 45 sec. Then my
Bluetooth token starts up.
Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port
1 disabled
Jan 14 00:13:00 sip2
2009 Sep 03
1
Originate calls with AMI.
Hello.
I've been trying to use the AMI to originate phone calls.
I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'.
So, the AMI interaction is:
> Action: originate
> Channel: SIP/zoiper
> Exten: yziquel
> Priority: 1
> Timeout: 30
> Context: internal
>
> Response: Error
> Message: Originate failed
>
> Event:
2003 Aug 15
1
DTMF SIP
Hello list,
my case is as follows:
SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729.
When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the
keypad on the phone.
As suggested by you, I need to configure the SIP1 with out band dtmf mode ,
what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238
? do I also need to make same kind
2005 Jul 18
0
why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi connected with asterisk manager
hello perl experts
i am working with "ast-rad-acc.pl" from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
i dont know why $cdr{'DNID'} and $cdr{'CALLERID'}
under 'sub send_acc {' are empty. i m successfully
connected with asterisk manager and when call i hangup
my perl application is getting that all other thing
are ok but i dont know why only
2009 Dec 01
0
Asterisk - Segmentation fault
Gentlemen,
Forgive me if I am posting at the wrong place!
I was going to test the "new" chan_ooh323 driver so I did install:
debian: Linux sip2 2.6.26-2-686 #1 SMP
dahdi-linux-complete-2.2.0.2+2.2.0
Asterisk SVN-trunk-r231692
Did enable chan_ooh323, everything compiled without any problems.
Hardware setup:
Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975)
X-Lite can
2004 May 20
0
budgetone problem on hangup
Hello to all.
I have a couple of budgetones connected to Asterisk
server. I can establish calls using budgetone with no
problem, but when I hang up a Budgetone, Asterisk
does not detect the hangup. It seems that the
communication goes on in spite of budgetone's hangup.
My sip.conf:
[general]
disallow=all
allow=ulaw
bindaddr=172.16.60.21
[sip1]
callgroup=1
pickupgroup=1
type=friend
2005 Feb 02
0
Speex pass through on SIP
Hi,
I've seen some answers to this on the mailing list archives but nothing
that seems like the right answer. What I want is for 2 SIP phones to use
speex to talk to each other through 2 asterisk boxes (linked over IAX2)
while only supporting ulaw on the asterisk boxes themselves.
I think a diagram will help ;)
SIP1 <--> *1 <--> IAX2 link <--> *2 <--> SIP2
I want
2005 Sep 22
1
Early Media with Asterisk
Hi :)
I hope someone has a hint concerning Early Media.
The situation:
My Asterisk is connected to small local carrier who works with several SIP
servers.
I traced some SIP headers and find something like this:
Via: SIP/2.0 UDP sip1.provider1.de
In the SDP part I found something like this:
o=- 2268929 0 IN IP4 sip2.provider1.de
c=IN IP4 sip2.provider1.de
If I send
2006 Apr 23
0
Re: Asterisk-Users Digest, Vol 21, Issue 132
Hi All
I want to do features as belows.
user ---> call ( from telco) --> asterisk ---> IVR -- SIP 1.
after that, SIP1 transfer to SIP2 (unattendant or attendant
transfer). i want to SIP1 hear stream sound data of call conversation between
user and SIP 2 (don't used call conference)
SIP3 want to hear stream sound data of user and SIP2 conversation,
can be press DTMF
2006 Apr 25
0
Re: Asterisk-Users Digest, Vol 21, Issue 132
Hi All
I want to setting as belows.
caller ---> call ( from telco) --> asterisk ---> IVR -- SIP 1.
after that, SIP1 transfer to SIP2 (unattendant or attendant
transfer). i want to SIP1 hear stream sound data of call conversation between
caller and SIP 2 (don't used call conference)
SIP3 want to hear stream sound data of caller and SIP2 conversation,
can be press DTMF