Displaying 20 results from an estimated 40000 matches similar to: "Disable transcoding"
2013 Dec 15
3
Why doesn't Asterisk try to prevent transcoding
Let's say I have two devices configured and the follow call scenarios occur.
[100]
disallow=all
allow=g722&ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with g722,ulaw,alaw,g729
101 dials 100 -> ulaw to ulaw is chosen
100 dials 101 -> g722 to ulaw is chosen
Ideally when 100 dials 101 ulaw would be chosen since it is the common
format.
2023 Jul 05
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello Michael,
you are referring to the following behavior - did I get it correctly?:
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and callee
(g722 <-> g711).
inbound works: call from provider: g711 -> asterisk drops g722 and
passes g711 to internal callee -> no transcoding.
As far as I know,
2023 Jul 05
3
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between
2003 Oct 27
1
Is transcoding a bad thing?
Hi there,
up till now I had this two-box setup in mind:
* no.1: public IP
* no.2: private IP, registers with no.1, serves a small office with
clients behind NAT
See we'd get something like this:
SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA
The codec of the SIP client (on the Internet) I don't have full control
over, that depends on the
2010 Feb 19
1
transcoding with TC400P
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[ 7.590966] Zaptel Version: 1.4.12.1
[ 7.590966] Zaptel Echo Canceller: MG2
[ 7.610963] zttranscode: Loaded.
[ 7.618969] wctc4xxp: tc400b0: Attached to
2010 Feb 16
1
CODECS: Best practice question: Avoid transcode when calling out?
What is the current best practice to avoid transcoding on an outgoing call
to a
party whose codec preference is not known in advance?
In other words, incoming calls are easy since codecs are negotiated from
least-known (the remote party) to most-known (my endpoint) and my codecs can
simply be preferred accordingly to match the remote.
Outbound calls seem harder. Our endpoints always negotiate
2014 May 12
1
new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted
transcoding is occurring on PSTN calls.
The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will
eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2,
CentOS 5.8) currently in production. Both systems are on VPS with public
IP addresses. Goals for the new system include: HD (g722) connections on
2003 Mar 02
12
Transcoding
Hello,
Does asterisk do transcoding when the call goes
through the system, codecs are the same but signaling protocol is changed.
example:
SIP with GSM ---> IAX with GSM
What quality destruction happen when I use transcoding? I know
this is not a concrete/precise question, but I would like to know how is
it in general.
What CPU performance is needed for transcoding 30 channels e.g.
from
2005 Feb 10
1
Codec passthrough patch for IAX
Hi there,
I had a problem, basically, I have 4 different types of end users
(gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider.
My provider supports all 4 codecs. The issue is then: When an incoming call
comes in, a codec is negotiated (usually ULAW), later on, when the extension
is dialed, we'll see we're doing GSM, and thus transcode. Here's an example
2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
otherparty realtime voice , but other party geting sip party's voice 1 sec
later (not
2006 Dec 07
1
Codec Selection in asterisk
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and
they all are able to register and make calls with no problem . My voip
carrier supports gsm as well as ilbc .. Server takes calls from sip phones ,
does call recording in between and forwards to voip carrier . My problem is
that half of my softphones use ilbc and rest use gsm and my provider
supports both gsm as well as
2006 Mar 31
1
transcoding g723 or g729 on asterisk
Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice versa.
I
2009 Sep 09
1
CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file
Good afternoon,
I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card.
The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty
file as you can see below...
CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw
2010 Jul 04
1
Asterisk for transcoding
Dear ALl
Can we use Asterisk for only for transcoding?. if yes how many concurent
call we can transcode with help of Astetrisk?
For this we only need to config SIP.conf or any other file too.
Thanks
Amit--
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2006 May 10
1
mg3000-r fxo gateway provides more feature to work with asterisk
Hi, every one
I'd like to introduce some new feature of our products.
mg3000-r fxo gateway provides more feature to work with asterisk.
1.play asterisk ivr with no interuption.
when the mg3000-r received call from co line, it wouldn't conect
instantly.instead, it start call to asterisk ivr first,when the ivr ready,
it connect the co line. this feature make user feel friendly.
2006 Jun 03
4
Meetme versus app_conference
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?
2003 Aug 14
1
Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding?
I have an Asterisk 0.4.0 install working with two grandstream budgetone
100 phones, gnophone, and kphone. This is a private network segment
(172.17.x.x), with the PBX configured on my outbound firewall which has
a public address (66.x.x.x).
- I can make calls between phones - all extensions are working.
- I can make IAX calls to IAXTEL. No problems (apparently gsm only)
- I can call SIP phone
2003 Dec 16
28
codec negotiation
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
My sip.conf is like that:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all
and cisco like that:
dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
2009 Sep 24
2
Digium transcoding card
Hi,
Given that the Digium transcoding card has no external connections
(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
Does such a beast exist, or is it likely to? Am I correct in assuming
that this is a Digium-only product, and there is no OEM equivalent
"generic" board out there that I could be investigating? It would be
such a shame to waste a PCI slot that
2023 Jun 30
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on originating
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on
outbound (called) channel and has no affect on calling channel. My
experiments and function documentation (which says “Media and codec
offerings to be set on an outbound SIP