Displaying 20 results from an estimated 5000 matches similar to: "How many gsm channels"
2007 Sep 20
4
Asterisk 1.2.24 simultaneous call limits.
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine that makes calls plays a message
(demo-intruct) upon the other machine answer. The machine receives the
calls just waits for 40 seconds
2012 Nov 22
3
monitoring asteriks
How can I monitor asterisk if all lines are registered etc?
I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc.
Usually I just restart asterisk and it solves the problem.
Is there an application that will email me if case any line looses registration with with asterisk?
Or any better solution!
--
Joseph
2006 Nov 20
2
Recording g729
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<font face="Helvetica, Arial, sans-serif">Before ordering I want to be
2007 Mar 02
1
DTMF detection problems on PRI channels?
I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks.
The application relies on a DTMF digit string sent by the phone
after the call has connected. This DTMF is detected by Asterisk
under the control of WAIT FOR DIGIT commands send from an AGI
processor over a FastAGI connection.
Usually the DTMF is detected without error, but on a significant minority
of calls, Asterisk is missing
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2007 Jun 28
2
Fax passthrough howto codec upspeed
Hello everybody,
Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax
When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT
2015 Jan 09
2
SEMI OFF-TOPIC - Fail2ban
2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>:
>
> Do you really want to detect "ChallengeSent"? That should occur also on
> legitimate login processes...
>
Hi , strange thing is that I still have not this asterisk in
production and I see many attempts Connection.
Now keep in mind that when a connection of authentication is
successful the
2007 Mar 01
7
IAX best practices
Hi guys,
I am planning to connect two Asterisk boxes that are currently running
in two different countries, using IAX.
I was wondering if anyone could provide me with some links or suggestion
regarding best practices in connecting two Asterisk in such way. I guess
many of you have already tried this, and already have some know-how
(what I should be careful about, what to avoid, etc...)?
2013 Oct 18
1
The codec can not support multi-thread ?
Hi! everybody:
We used opus-codec for a VOIP gateway. The GW is running at a UBUNTU server.
The opus stream is transcoded to G711 pcmu stream.So there are many opus
codecs running simultaneously.
We noticed that if there more than 5 streams in. the voice then has
notisable glitchs.More streams in, worse voice got.
Then we write test code for opus-codec which encode a .pcm file
simultaneously.
2007 Mar 05
4
TC400B
Anyone tried the digium TC400B transcoding card? What are your opinions?
Thnx
2010 Apr 19
1
zapg723toslin did not update samples
hello
i am using a TC400B transcoding card, and sometimes when a G723 call is
coming in, that is getting transcoded to G711, the CLI is flooded with
..
[Apr 19 17:39:32] WARNING[3336] translate.c: zapg723toslin did not
update samples 720
[Apr 19 17:39:33] WARNING[3336] translate.c: no samples for zapg723toslin
[Apr 19 17:39:33] WARNING[3336] translate.c: zapg723toslin did not
update samples
2006 Feb 20
1
g729 quality at GSM bitrates
Greetings all,
I'm trying to improve the codec selection on a few of the asterisk boxes we
have to keep the g729 licences free for calls from ATAs that don't support
anything apart from g711 and g729. GSM seems to offer noticably inferior
call quality (at least when using a softphone + decent headphones), but it's
about where I want the bitrate to be.
I know there are lots of Speex
2008 Feb 07
5
Two Leg CDR
Hi all,
i am wondering if i can make two leg cdr in mysql cdr table.
1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table.
2nd Leg : The CDR of carrier for the example if i send call like
exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP)
I this cause i can get the accrue duration of call because currently we are facing some call missing not coming
2006 Oct 13
1
3way calling / codec problem
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs!
Any help is greatly appreciated!
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UAC ASTERISK UAS | ASTERISK UAC
PROVIDER
g711 <----------------------> g711
2007 Sep 10
5
Asterisk Manager API - Originate command
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command very fast. Anyone know if there is a limitation? Thnx.
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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,
2007 Jul 02
4
Help. Cannot compile version 1.4.6 with the following error
Hi all,
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c:2309: warning: implicit declaration of function
`pri_keypad_facility'
chan_zap.c: In function `pri_dchannel':
chan_zap.c:9292: structure has no member named `call'
make[1]: *** [chan_zap.o]
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
2007 Apr 09
3
Upgrade 4 to 8 Analog Lines Question
Hello
I have an office with a T1 that provides 4 (out of 8) analog PSTN
lines thru an adtran board. I want to add 4 more analog lines.
Currently I have a Digium TDM40B. I'm wondering what the best
upgrade path is, where I define 'best' as the solution that
is most likely to work without problems (like interupt conflicts)
and work with my current echo tuning .
I see my purchase