[asterisk-users] Problems with ulaw/g729 translation

2010-07-05 Thread Felipe Neuwald 
Dear Folks,

I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is
simple: connection to the PSTN directly via SIP, using g729 codec, and
connection to the softphones (X-lite 3.0 build 56125) trought local network,
using ulaw codec.

Sometimes, I got messages like:

[Jul  1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported
SDP media type in offer: image 65344 udptl t38


And then a lot of messages like:


[Jul  1 15:27:00] WARNING[26549]: translate.c:274 ast_translator_build_path:
No translator path from alaw to unknown


That's stopping the phone system. When I got the messages, I can't make or
receive calls. Then, a few minutes later (or when I stop and start
asterisk), the phone system back to work again.


Some confs and system status:


sip.conf:


[1050] ; THAT'S A SOFTPHONE

type=friend

host=dynamic

callerid=Softphone 1050

secret=

context=call-center

disallow=all

allow=alaw

allow=ulaw

dtmfmode=rfc2833

canreinvite=yes

nat=no

qualify=yes

call-limit=1

allowtransfer=yes

insecure=no

promiscredir=no

useclientcode=no

videosupport=no


[7600] ; THAT'S PSTN CONNECTION

username=7600

type=friend

secret=

qualify=no

port=5060

nat=yes

mailbox=7...@default

host=dynamic

dtmfmode=rfc2833

context=out

canreinvite=no

callerid=7600

disallow=all

allow=g729


[sipgvt] ; THAT'S PSTN CONNECTION

username=1121317600

type=peer

secret=

port=5060

insecure=very

host=gvt.com.br

fromuser=1121317600

fromdomain=gvt.com.br

dtmfmode=rfc2833

context=in

disallow=all

allow=g729


neuwald01*CLI> g729 show licenses

3/8 encoders/decoders of 30 licensed channels are currently in use


Licenses Found:

File: G729-x.lic -- Key: G729- -- Host-ID: x -- Channels: 30
(Expires: 2030-06-07) (OK)

neuwald01*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INTBINARYHEX   TYPE   NAME   DESC

  1 (1 <<  0)  (0x1)  audio   g723   (G.723.1)
  2 (1 <<  1)  (0x2)  audiogsm   (GSM)
  4 (1 <<  2)  (0x4)  audio   ulaw   (G.711 u-law)
  8 (1 <<  3)  (0x8)  audio   alaw   (G.711 A-law)
 16 (1 <<  4) (0x10)  audio   g726aal2   (G.726 AAL2)
 32 (1 <<  5) (0x20)  audio  adpcm   (ADPCM)
 64 (1 <<  6) (0x40)  audio   slin   (16 bit Signed Linear
PCM)
128 (1 <<  7) (0x80)  audio  lpc10   (LPC10)
256 (1 <<  8)(0x100)  audio   g729   (G.729A)
512 (1 <<  9)(0x200)  audio  speex   (SpeeX)
   1024 (1 << 10)(0x400)  audio   ilbc   (iLBC)
   2048 (1 << 11)(0x800)  audio   g726   (G.726 RFC3551)
   4096 (1 << 12)   (0x1000)  audio   g722   (G722)
  65536 (1 << 16)  (0x1)  image   jpeg   (JPEG image)
 131072 (1 << 17)  (0x2)  imagepng   (PNG image)
 262144 (1 << 18)  (0x4)  video   h261   (H.261 Video)
 524288 (1 << 19)  (0x8)  video   h263   (H.263 Video)
1048576 (1 << 20) (0x10)  video  h263p   (H.263+ Video)
2097152 (1 << 21) (0x20)  video   h264   (H.264 Video)
neuwald01*CLI> core show translation
 Translation times between formats (in milliseconds) for one second
of data
  Source Format (Rows) Destination Format (Columns)

  g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
g722
 g723-   ---- -- -- ---
   -
  gsm-   -222 21 37 --2
   -
 ulaw-   2-12 21 37 --2
   -
 alaw-   21-2 21 37 --2
   -
 g726aal2-   222- 21 37 --2
   -
adpcm-   2222 -1 37 --2
   -
 slin-   1111 1- 26 --1
   -
lpc10-   2222 21 -7 --2
   -
 g729-   2222 21 3- --2
   -
speex-   ---- -- -- ---
   -
 ilbc-   ---- -- -- ---
   -
 g726-   222    2 2    1 37 ---
   -
 g722-   ---- -- -- ---
   -

Any idea?

Thanks,

Felipe Neuwald.
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[asterisk-users] Asterisk and HPC Cluster

2007-09-17 Thread Felipe Neuwald 
Hi all,

does anybody have implemented asterisk on a Beowolf cluster (HPC - High 
Performance Computing) using Message Passing Interface (MPI)?

Thanks,

Felipe Neuwald.

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[asterisk-users] Round Robin Queue

2007-01-10 Thread Felipe Neuwald 

Hi Folks,

I implemented an Asterisk 1.2.10 on a Debian GNU/Linux, and I have
implemented a round robin queue (and a memory round robin queue too).

Here I have one simple problem:

- agent 1 (busy)
- agent 2 (busy)
- agent 3 (free)

When someone call to my queue, the action of the queue is this:
call agent 1, then call agent 2, and then call agent 3, that is free and
finally ring. There is someway to my queue only call free agents?

Thank you,

Felipe Neuwald.
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[asterisk-users] asterisk (FreePBX) and queues

2007-01-05 Thread Felipe Neuwald 

Hi folks,

I'm using a fewestcall queue here, and I'm having the follow problem:

I have 3 static agents in my default queue:
2001
2002
2003

User 2001 and 2002 are logged in, but 2003 are logged out. When someone call
to my default queue, the queue try to ring 2003 (that isn't logged). There
is some way to the queue only ring logged users?

Here is my show queue:

zeus*CLI> show queue 100
100  has 0 calls (max unlimited) in 'fewestcalls' strategy (4s
holdtime), W:0, C:3, A:3, SL:0.0% within 0s
  Members:
 Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet
 Local/[EMAIL PROTECTED]/n (Unknown) has taken 1 calls (last was 346
secs ago)
 Local/[EMAIL PROTECTED]/n (Unknown) has taken 2 calls (last was 195
secs ago)
  No Callers

Thank you,

Felipe.
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