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=xxxx
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=xxxx
qualify=no
port=5060
nat=yes
mailbox=7600 at 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=xxxx
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-xxxxx.lic -- Key: G729-xxxx -- Host-ID: xxxxx -- 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.
INT BINARY HEX TYPE NAME DESC
--------------------------------------------------------------------------------
1 (1 << 0) (0x1) audio g723 (G.723.1)
2 (1 << 1) (0x2) audio gsm (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) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) 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 - - 2 2 2 2 1 3 7 - - 2
-
ulaw - 2 - 1 2 2 1 3 7 - - 2
-
alaw - 2 1 - 2 2 1 3 7 - - 2
-
g726aal2 - 2 2 2 - 2 1 3 7 - - 2
-
adpcm - 2 2 2 2 - 1 3 7 - - 2
-
slin - 1 1 1 1 1 - 2 6 - - 1
-
lpc10 - 2 2 2 2 2 1 - 7 - - 2
-
g729 - 2 2 2 2 2 1 3 - - - 2
-
speex - - - - - - - - - - - -
-
ilbc - - - - - - - - - - - -
-
g726 - 2 2 2 2 2 1 3 7 - - -
-
g722 - - - - - - - - - - - -
-
Any idea?
Thanks,
Felipe Neuwald.
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Eyal Goltzman
2010-Jul-05 17:57 UTC
[asterisk-users] How to change the IP in the SIP contact header
Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: "Polycom" <sip:100@ x.y.z.w>;tag=as7435100b To: <sip:144@ a.b.c.d > Contact: <sip:100 at 10.100.101.107> Call-ID: 08116cf06661dc091de10c1b3315d2f7 at 84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100705/cee7cf4b/attachment.htm