Displaying 20 results from an estimated 4000 matches similar to: "How to change the packet size"
2004 Aug 31
2
Asterisk codecs and packet size
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect remote sip clients with 24kb bandwidth
lines and I'm using a licences g729 codec but because I can't increase
2004 Dec 10
1
Doubts regarding g726 - 16 bits setup
Hi all,
I would like to make a call using the asterisk IAX
with g726 - 16 bits codec.
How could I configure it in the iax.conf file.
Do I need to modify the file like this?
.
.
disallow = all
allow = g72616k
.
.
I have tried it but it hasnĀ“t worked.
Thanks in advance and best regards
Guild
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2005 Jun 09
4
Lingo(.com) and Asterisk
Hello,
A long Google search didn't turn any clear answer. Does somebody use
Asterisk in combination with Lingo?
Thank you,
Bas
2017 May 12
3
pjsip: asterisk can't decide which codec to use
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone extension -> asterisk -> provider A -> provider B -> asterisk.
Asterisk initially sends
2007 Mar 08
1
Issues with a Linksys SPA 2102 and asterisk
Topology:
analog_phone-----SPA2102-----Navini_Wireless_Router------ISP------Asterisk
A ping against the asterisk server shows aprox 145ms roundtrip.
128kbps upstream
512kbps downstream
g729a as codec
signal quality of the navini router: 100%
The ATA operates correctly in every form, however sometimes when
someone is talking to me (the other person is at pstn) and then I
start talking the other
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
2008 Dec 05
2
async agi question
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
2010 Aug 04
1
Asterisk not working with Festival
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten => s,1,Answer
Exten => s,n,SayDigits(?1?)
exten => s,n,Festival(hello john)
exten => s,n,Hangup
I use call files to
2004 May 04
1
Probs with oh323 driver: audio only in 1 direction
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus)
and oh323-0.6.0
Here are my config's
##############
# modem.conf #
##############
2009 May 22
3
No response to our critical packet problem
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect
2005 Dec 13
4
compaq r4000 32 bit and 64 bit install
Gents,
I first installed the 32 bit version (as you might have read) of centos 4.2,
after downloading x86_64 the install process/progress is about the same.
going VERY slow...
The screen for formatting the partitions took a LONG time
(3 partitions 10 GIG, 2 GIG swap and 50 GIG).
While on that screen and moving the mouse it lagged big time and
could not keep up.
Anyway just reporting that so far
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)
==
2010 Aug 02
6
Codec negotiation : expecting G726, getting G711a (alaw)
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.
The SIP peers are both defined as :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
This is the
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
2014 Jan 23
1
mixmonitor extension
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---------------------------------------
Marek Cervenka
=======================================
2006 Apr 11
2
G726-40 required - Help!
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager already offered this to
the customer and now i do not know how to do it...
Thanks a lot in advance,
2005 Jun 02
2
asterisk sipura and g726 codec
With sipura (I tried this with both the 3000 and 841) set to prefer
the g726-32 codec, a call from the sipura to asterisk will use g726.
Asterisk sip.conf has:
disallow=all
allow=g726
allow=gsm
allow=alaw
When the call is from asterisk to the sipura, asterisk will not use
g726. It ends up using alaw. I usually use stable but I tried this
with head too, and same thing happens.
Anybody know how
2008 Jan 15
3
Meetme recording
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
********************************************************************
This email and any attachments
2005 Mar 16
2
Dial multiple extensions, but different variables/timeouts
Hi everyone,
I'm wondering I would accomplish the following: I want to dial several
SIP extensions simultaneously, HOWEVER, for different times (say ext
10 for 15 sec and ext 11 for 30 sec), and potentially with different
headers (such as ALERT_INFO) and codecs for each extension. Obviously
whoever picks up first gets the call. After the longest timeout
expires (30 sec in this example) I want
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723