Displaying 20 results from an estimated 100 matches similar to: "Modifying cid.cid_name in app_parkandannounce.c"
2006 Mar 14
3
Attended Transfer - transfer timeout, how to change?
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have changed the extension for this number to timeout
at 60 seconds, but that seems to make no difference.
--
2006 Nov 15
2
some questions about atxfer usage
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
"transferer". Is there any possibility to recover the call before the
timeout of 15 seconds expires?
I mean, I would like
2004 Aug 11
0
Inband announcement of parking slot from app_parkandannounce?
I'm trying to use Asterisk app_parkandannouce to build a global parking
pool from within a couple of Norstar PBXes. Right now I can blind transfer
calls into the parking lot, but the slot announcement relies on calling back
the 'transferee' after the call is parked and I can't pass enough callerid
data out from within the PBX to be able to route the call back in (ie. no
PRI
2006 Dec 20
0
asterisk run on vxworks for hardware pbx
Hi
My hardware PBX run asterisk on vxworks,Because the vxworks not support
perl.
Now I want to add a callback function to my pbx.
now it can store Caller and Called party numbers in queue when Called party
is busy
Then I malloc a new ast_channel to call.It is should use
ast_get_channel_by_exten_locked() or ast_channel_alloc() ,
my program as follow,But it isn't work, anyone know how to
2007 Jun 26
1
CDR Records "s" as dst
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of "s" as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten =
2010 May 06
2
problem with trustrpid
Hi everyone,
I am trying to figure out the behavior of trustrpid
Basically its not behaving the way I expected it to or maybe I am
missing a configuration option or something else.
When a call from a phone is sent to the * box it has the following sip
headers:
From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476.
Remote-Party-ID: "Cloutier"
2005 Feb 14
4
Asterisk-H323
Greetings,
I have a problem making a call from Asterisk to Cisco H323 PSTN gateway
using H323 channel. I can call but there are no sound in both way. If I call
H323 gateway directly from SJPhone I have no problem with sound.
Any advice are welcome.
Thanks in advance.
2007 Aug 27
3
voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my
problem is the following one:
when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown
in the asterisk CLI and caller and callee can hear each other when
2005 Jan 13
1
SCCP questions
Hi!
I have two, not too related questions:
- the probably simpler one: if anyone can help me out using a Cisco
7905G with chan_sccp? I did already managed to get it working with a SIP
image, I'd just like to see it work with this one as well. It's probably
something I screw up with the configuration, as the phone registers,
only I don't get any lines with it, although I have it
2004 Jun 30
3
Bugfix for CVS-HEAD-06/26/04-21:56:45
Hiya,
I sent this bugfix to the asterisk-dev mailing list, and modified it as I
noticed side effects, but now it appears to be finished. Nobody seemed to
notice it there, so I thought I'd post here, as it seems to be something
that will be needed as people update to the latest CVS version. So...read
on :)
Ted
programmer_ted@hotmail.com
P.S. Read to the very end. The original bugfix
2006 Apr 09
0
Realtime oracle compiling problem
I can'T compile my oracle realtime library any more i updatet the svn
today and now i tried to recompile my oracle realtime driver and now it
gives me that errors:
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/include/oracle/10.1.0.4/client -c -o res_config_oracle.o
res_config_oracle.c
res_config_oracle.c:53: warning: data definition has no type or storage
class
res_config_oracle.c: In
2009 Oct 14
1
ChanSpy on asterisk 1.6
I have read about that on asterisk 1.6, there will be a parameter "o" (Only
listen to audio coming from this channel), I have tried, but I still get
inbound and outbound audio from the spied channel.
Has anyone used this feature? Is it working? Is there any work-around?
I will like to only spy the outbound audio from a channel, I dont want to
hear the incomming audio of that channel.
I
2004 Aug 19
1
Inband announcement of parking slot from app _parkandannounce?
Couldn't see the forrest for all the fascinating tree-like applications that
are out there:
For future reference, see:
http://www.voip-info.org/wiki-Asterisk+call+parking
:-)
-----Original Message-----
From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca]
Sent: August 11, 2004 1:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inband announcement of parking slot from
2004 May 03
3
Error building asterisk-0.9.0
I am trying to build asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0
machine.
I have followed the instructions to build asterisk.
Building zaptel and libpri seemed to go well (lots of messages but
nothing that indicated an error)
However, when I do the make clean ; make install for asterisk-0.9.0
after running for sometime I get the following:
gcc -shared -Xlinker -x -o app_senddtmf.so
2006 Mar 16
0
SCCP problem with ATA188, Asterisk@home and chan_sccp
Hi,
This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ).
I'm trying to get SCCP ATA188s to run with Asterisk.
The Asterisk box uses the latest Asterisk@Home image (Version 2.6).
I have compiled and
2003 Apr 15
4
call announce?
using a zap fxo and zap fxs card how can I set up caller announce? like
this.
1 call comes in and a prompt asks the called to identify themselves.
2 the system would then put the caller on hold and pick up the FXS and
play the message for the users prompting them to hit 1 to accept the
call and have it connected or hit 2 to dump the live caller to
voicemail.
Can this be done with *
Dave
2007 May 23
0
Problems compiling res_config_mysql (asterisk addons)
Hello All:
I'm having some difficutly getting res_config_mysql from the 1.4.1 addons
package to compile ( I need it for Realtime)
First of all, when I make everything appears to compile ok with no errors
however the res_config_mysql doesn't get compiled. So I tried "make
res_config_mysql" and a whackload of errors starting with the following:
# make res_config_mysql
gcc -g
2005 Jan 04
0
cid_num with Asterisk CVS 1.0.12
Hello,
How can I access caller's number with Asterisk CVS 1.0.12?
In new version there are structure cid with field cid_num. And in 1.0.12
only callerid field which is equal to cid_name.
I also tried to get it from chan->cdr->src but this is also the same as
cid_name or callerid.
Mindaugas Kezys
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2007 Jul 12
0
No subject
static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
{
manager_event(EVENT_FLAG_CALL, "Dial",
"Source: %s\r\n"
"Destination: %s\r\n"
"CallerID: %s\r\n"
"CallerIDName: %s\r\n"
"SrcUniqueID: %s\r\n"
"DestUniqueID: %s\r\n"
"CDRUserfield: %s\r\n",
src->name,
2009 Oct 05
3
Questions about app_jack.c
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular &
1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to