Displaying 20 results from an estimated 3000 matches similar to: "Digium G.729 codec modules updated"
2005 Sep 05
9
Asterisk Follow ME
Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.
When call goes via IAX and calling part "accept the call" on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.
If anyone faced with such problem please let me know. I need to know
whether it's bug or just configuration
2005 Oct 05
3
SIP Attended Transfer using REFER and Replaces: headers
hey all,
am wondering if anyone has successfuly done a SIP attended transfer using
the REFER method (after an INVITE obviously) and the Replaces: header.
we're writing our own SIP UAC and the asterisk code seems to support it,
but we're not really sure if this is so.
we plan on the following call flows:
1. incoming call from exten 1111 is sent to our UAC with Dial()
2. our UAC makes
2006 Jan 20
5
iDEFISK (mac iax2 softphone) release
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also put a newer version of the windows and linux
versions online.
Let us know how you feel about it, a more mac look (brushed metal) is
coming.
2005 Oct 10
2
DTMF Question (misunderstood '*' button)
Hi all!
I'm experimenting a strange problem in my Asterisk PBX:
I've got an Asterisk pbx in the office: I dial an external number; the dialled
number answers me correctly, but as soon as I press the '*' button (i.e. to
navigate through the menus or to enter a password) my Asterisk box put me on
hold.
(CLI transcription follows:
-- Executing
2001 Oct 19
2
wine 20010824 and quake
i have quake v1.06 installed and running fine under windows. however,
running it in wine gives a bunch of errors. see below:
prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath
fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31
!
prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath
fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.
2005 Oct 14
5
sip accounts
hi, i facing a problem here. in my sip.conf, i specify a account like this,
[1234]
type=friend
context=from-sip
username=1234
secret=1234
nat=no
canreinvite=yes
dtmfmode=info
mailbox=1234@default
disallow=all
allow=ulaw
so i am able to login with username 1234 and password 1234
but ther weird part is, i can also register as any number (account)
without having to specify in sip.conf. thus
2005 Jul 01
3
pattern matching based on callerid, not working
according to the wiki, I should be able to do this:
exten => _9./3003,1,Set(CALLERID(number)=2814444443)
exten => _9./3004,n,Set(CALLERID(number)=2814444444)
exten => _9./3005,n,Set(CALLERID(number)=2814444445)
exten => _9./3006,n,Set(CALLERID(number)=2814444446)
exten => _9.,n,Dial(SIP/${EXTEN:1}@mycarrier,30,wt)
and have the correct calleridnum's set for each extension based
2006 Jun 12
2
Attended transfer and queue
It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active.
Does anyone know any workarounds for this problem?
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2006 Mar 28
4
RTP frame size location?
Google has given me too many responses, so I'll ask the list:
Where in the Asterisk rtp source code can I find the default
millisecond frame size? I've looked around for obvious pointers, but
it's not clear. I'd like to "force" my Asterisk server to use a
certain frame size all the time. (Of course, ideally I'd like to
prefer or even force that frame size in a
2006 Jan 31
5
Queue() with timeout=0
Hello,
i've recently switched over from 1.0.9 to 1.2.3.
I've experienced some (to me) weird behaviour.
This is the config for an example queue.conf:
[654]
wrapuptime=30
timeout=20
strategy=ringall
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
music=default
monitor-join=yes
monitor-format=
2006 Jun 27
5
WebPhone
Hi,
someone know a good webphone, possibily a free one
Thx
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2006 Mar 02
3
Native music on hold - Error
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have
[native]
mode=files
directory=/var/lib/asterisk/moh-native
And in sip.conf I have
musicclass=native
When I put call on hold this is what I get at CLI.
-- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack
2006 Apr 07
2
Announcing Astmanproxy 1.20
Greetings everyone,
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
allows you to communicate with multiple Asterisk boxes from a single point
of contact using a variety of I/O formats, now including support for
XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format.
Astmanproxy is
2004 Sep 19
1
RE: [Asterisk-Dev] Hardware details for the Digium TDM400P
asterisk-dev-bounces@lists.digium.com wrote:
> I have a DSP based system that is working on a four port FXS system
> using a 200MHz arm processor.
Well.. since we are talking about this topic I owe you guys notes of my
experience
with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch
etc.).
We made a Linux distro and compacted it into 32MB flash. Installed asterisk
and
2005 Oct 10
3
Billing/SPA-841/CDR Log
Hi list,
I have a couple of questions related to asterisk billing and the
generation of cdr logs. I've searched the wiki but have not found my
answers, hopefully you guys can help.
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the
records created and it seems to only generate it at the time the call
is
2006 Feb 27
5
res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind
2006 Mar 22
2
Pickupexten not working
Hi group. I have huge problem. My pickup exten #8 isn't working.
This is what I have configured.
pbx*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 #8
In sip.conf I have
callgroup=2
pickupgroup=2
For called party and same for person that is trying to pick up the call.
The person that is trying
2006 Apr 05
3
queue issue
Hi,
I have several queues configured at my call center for different support levels.
Today, something weird happened:
- A client called queue 1 and was answered by an agent
- The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf
- The user transferred the client to another Queue, by using the second channel and the XFer key of her
2006 Jun 13
8
IAX2 Vs SIP cpu load
Hello
Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads?
Also, does Asterisk support and use multiprocessor architectures, such as Xeon?
?
Regards
Jon
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2006 Mar 09
4
IVR woes
Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working.
Firstly the asterisk version is:
Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC
Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily