Displaying 20 results from an estimated 200 matches similar to: "G729a codecs + Asterisk 1.4.11"
2004 Mar 30
1
G726 not working ?
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced".
When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
[format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
==
2007 Oct 08
2
inbound call voip providers
Hello:
I want to have a local telephone number that, when the people calls this
number (via mobile or normal PSTN), the voip provider stablishes a SIP
session to my asterisk box.
It is possible?
If yes...
What providers have this service in Europe?
It is difficult to configure and get things working ok?
Will my asterisk box see the mobile or normal PSTN phone# that is calling the
number
2007 Jan 26
3
International Carriers
Hello everyone!
I 've looking for carriers which can terminate my international calls.
They must accept payments from Argentina and give me interconection to
my Asterisk. I'd appreciate your help or recomendations.
Regards.
--
Facundo Ameal.
fameal<at>gmail<dot>com
Linux User #395088
Share your knowledge, use free software.
2007 Nov 01
1
AsteriskNOW and TDM800P
Hi all
I sold new TDM800P card with 8 FXO ports, someone know if can be use
this card on AsteriskNOW or trixbox?
What can i do for use this card?
Thanks.
----------
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
rcm at fonetglobal.com
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
+ 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
-------------- next part
2004 Jan 16
4
G.723.1 codec
Hi,
Want to do some experiments with the G.723 codecs - where can I download the
723 source code for Asterisk?
I know there are some ongoing discussion regarding patents and license fees
for the g.723 but I have some hardware on which I only have the 723 and need
to test it privately.
Thanks!
Dan
_________________________________________________________________
Use MSN Messenger to send
2008 Nov 13
0
Problems with Licensed g729a codec from Digium
Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |------------- <SIP/uLAW> --------------| ASTERISK
|-------------- <IAX2/g729> ------------|VoIP/ISP
The g729 codec has been registered successfully and appears to be
detected by Asterisk
(NOTE: I have changed what I thought might have
2006 Apr 10
0
WG: G729a error
Somebody can say me what i can do that the g729 is working?
_____
Von: Ren? Enskat [Teamware GmbH] [mailto:ren@teamware-gmbh.de]
Gesendet: Montag, 10. April 2006 10:21
An: 'asterisk-users@lists.digium.com'
Betreff: G729a error
when i load asterisk i got this error and cant start * with the g729
codec:
Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18
2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
Please help.
Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=xxxxxxxxxxxx
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes
Asterisk 2 sip.conf
GNU nano 1.3.12
2005 Aug 14
2
Bigger problems than ogg
Ok,
After following BJ's advice and removing ogg.so I then got a
pbx_realtime.so error in the same fashion. I removed that file, and
then the next and then the next as you can see in the log below.
I think something is not right. duh here is my sign..lol...but I am
not sure even where this ast_register_file_version flag is in a config
file or what step I have missed. I am doing a VOIP only
2011 Jun 29
1
No audio format found to offer.
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729
However, the Dial application gives the following error:
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP -> SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give
2005 Mar 21
2
G726-16 passthrough...
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been working fine
for me, but it does not. G726-32 does work great however, but it's like
Asterisk doesn't
2007 Oct 23
0
Internal Data Stream Error
Hello again,
I am using mix monitor and the majority of the sound records perfectly.
I then get a "Internal Data Stream Error" near the end of the sound
file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs
and an example dialplan entry is ;
; phone line phone1
exten => phone1,1,Answer()
exten => phone1,2,MixMonitor(test.wav|av(0)V(0))
exten =>
2006 Feb 14
1
voicemail recording format
Dear asterisk users,
I am presently playing with an asterisk@home. I am trying to find the
best codec solution for my voicemail records. I want to use ARI
(Asterisk Recording Interface) to read the messages. I first used the
default wav encoding that was not appropriate because my navigator does
not handle wav mime types correctly and I have difficulties playing wav
files in my basic linux
2004 Jul 29
0
G.729 between Zap and SIP
Hi,
I have licensed the digium G.729A codec. But for some reason incoming and
outgoing calls will ALWAYS use G.711a. When I force my phone to only accept
G.729 then an incoming call from ZAP goes straight to my voicemailbox as the
phone doesn't accept the codec Asterisk wants, even if I force it in
sip.conf.
Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ?
The
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL,
Any clues or tips for the following gdb messages.
[root@localhost asterisk]# uname -a
Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct
29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux
localhost*CLI> show version
Asterisk CVS-HEAD-09/22/04-11:19:09 built by
root@localhost on a i686 running Linux
[root@localhost asterisk]# gdb asterisk core.13089
GNU gdb Red Hat Linux
2004 Jul 30
0
G.729 <-> ZAP ?
Hi,
I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card.
Incoming calls and outgoing calls between my cisco and my SIP phone works
fine on G.729. Recording messages in the asterisk voice-mailbox also works
fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have
licensed the digium G.729A codec.
When I connect my ISDN PRI to my Zap card and I call
2005 Mar 20
1
I cannot use G711 (ulaw|alaw)
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
I do not understand why because my Asterisk box load these codecs properly!
Does somebody
2003 Apr 04
2
chan_h323 problems....
I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc.
I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs
pwlib
openh323
gnugk for h.323 gatekeeper