Displaying 20 results from an estimated 20000 matches similar to: "Transfer through AGI"
2006 Dec 12
1
AGI problema
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<font face="Verdana">Hi all. I've written a AGI in C language.
2007 Aug 17
0
analog lines running agi on hangup question
I have the following dialplan.
Everything seems good except for one thing.
If the background message is playing and the user hangs up and does not
press a digit
how do I run an agi on that event.
I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never
called.
I am using 1.4.10
thanks,
Jerry
---------------------------
[smvoice-analog]
exten => s,1,Wait(1)
exten =>
2008 Aug 07
1
outgoing call file and agi detect busy
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401 for
smvoice_callprogress at smvoice-dialout:1 (Retry 1)
-- Got SIP response 486 "Busy" back from 192.168.1.161
2009 Oct 28
1
Asterisk 302 Moved Temporarily
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Hello,<br>
<br>
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE
2005 Jan 25
0
OH323 Cisco Transfer Key
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-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
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Hi
2005 Aug 02
0
app_rxfax errors
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Up until today, I have had no issues with receiving faxes in *. One
change I made was that I now have the incoming
2010 May 26
1
Getting "ghost" transfer or music on hold
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<font face="Tahoma">Hi Everybody,<br>
<br>
I´m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ...
In some calls, i get an atxfer or musiconhold in the middle of
2007 Jan 13
0
MHII.OB(MARSHALL HOLDINGS INTERNATIONAL INC.) this special for you
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<HTML><HEAD><TITLE>MHII.OB this is really amazing company that you always dreamt</TITLE>
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<head>
Following a meetingHe told the BBC there would be no UN troops. He told the BBC there would be no UN troops. <br>
<meta
2011 May 04
1
asterisk 1.4.35 to 1.4.41
Under 1.4.35 I get this message printed MANY times
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000 (g722)(4096)
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000
2009 Jun 04
2
broken pipe in perl agi
Hi gang,
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for line
in use that way. The AGI works fine from a prompt, but returns the dreaded
"utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I
try to run it from the dialplan. Here is my dialplan snippet;
2010 Nov 27
2
Preserve CallerID on transfers
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<font face="Arial">Hi, it´s possible to mantain the original
2010 Sep 07
1
Solving the CDR mess of attended transfers
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<font face="Arial">Is there a way to solve the mess on CDR caused by
CDR
2011 May 17
1
Question on AMI
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically "core show
channels concise"
"sometimes" I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF
]'
I'm not
2009 Jul 06
0
Iax trunk quality
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style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi,
<br>
<br>
I try to find a solution for this problem : <br>
2008 Sep 26
2
server and 2 uniden phones no ringing
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but that made no
difference after restarting.
Very simple network. server, linksys router and 2 phones.
2009 Jul 02
1
AGI Transfer?
I've been trying to get an AGI "transfer" to work for several weeks now. It
isn't error-ing out, but it isn't working either.
I can't use "dial" in this case due to what I'm trying to accomplish.
Does an AGI Transfer actually work?
-= Info about application 'Transfer' =-
[Synopsis]
Transfer caller to remote extension
[Description]
2007 Apr 19
1
Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine.
However, using outgoing call files the CS1000 is hanging up after I answer the call.
I dont know why?
Thanks, for any assistance.
Jerry
my sip.conf entry is:
[Nortel]
type=friend
dtmfmode=rfc2833
username=XXXXXXXXX
disallow=all
allow=ulaw
allow=alaw
2003 Jul 14
0
Cisco 7960 Transfer Call drop problem
Hi,
I'm having problems with transfer from an analog line via a X100p and Cisco
7960's running SIP.
With an attended transfer the a call comes in, I transfer it to another
7960, they answer I announce the call, press transfer again, the two parties
talk for 1-2 seconds then the analog line drops, though the Cisco phone is
not aware of this, i.e. nothing on the screen changes. The
2004 Aug 06
0
toner cartridges
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<title>fgfg</title>
2006 Mar 19
0
Transfer to specific park number
Hi
I'd like to allow users to transfer a call to a specific park number. This
way, the receptionist can tranfer a call park for ext 100 at park number
7100 etc...
It seems like this should be fairly simple using the Park(ext) app but it
doesn't work for me. No matter what I extension I use, the system just
picks the next available park number. I've simplified my dialplan for