[Home]

Summary:ASTERISK-05841: I don't want ilbc, i just want G.711...
Reporter:Jason Chan, Hong Kong (jason_polaris)Labels:
Date Opened:2005-12-14 08:59:30.000-0600Date Closed:2011-06-07 14:02:57
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/CodecInterface
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Hi there,
I am writing to ask about how to fix the codec to G.711 ONLY.
Actually what I am doing is, try to use DTMF when the POTS phone call has directed to Asterisk via Planet VIP-450 FXO Port, but this gateway just simply doesn't support RFC2833 nor SIP-INFO. The only method I can use is Inband DTMF. I know it only support G.711, but I DID disallow others and make it work only with G.711. But the problem is, although I disallow all other codecs, ilbc still itching me...
[extensions.conf]
[852]
username=HKGW
serect=blah
type=friend
host=dynamic
nat =yes
canreinvite=no
disallow=all
disallow=ilbc
allow=ulaw
dtmfmode=inband

(P.S. I don't use REINVITE simply because I need the asterisk to be a media gateway cause the gateway is inside NAT behind the Asterisk)
Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got such messages:

Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec ilbc. Use RFC2833
Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?

How come!? I DID DISALLOW them, but it keeps bugging me....

=====
192.168.2.3      852         79f9e0-c0a8  00101/00001  ulaw  No       Rx: ACK
1 active SIP channel
*CLI> sip show channel 79

 * SIP Call
 Direction:              Incoming
 Call-ID:                79f9e0-c0a80203-13c4-3a53f3e1-bbfcaf8-3fcf@192.168.2.3
 Our Codec Capability:   4
 Non-Codec Capability:   0
 Their Codec Capability:   261
 Joint Codec Capability:   4
 Format                  ulaw
 Theoretical Address:    192.168.2.3:5060
 Received Address:       192.168.2.3:5060
 NAT Support:            Always
 Audio IP:               192.168.2.1 (local)
 Our Tag:                as737358ce
 Their Tag:              3a53f3e1-bbfcafe6d5c
 SIP User agent:
 Username:               852
 Peername:               852
 Original uri:           sip:8888@192.168.2.3:5060
 Caller-ID:              elite
 Need Destroy:           0
 Last Message:           Rx: ACK
 Promiscuous Redir:      No
 Route:                  sip:8888@192.168.2.3:5060
 DTMF Mode:              inband
 SIP Options:            (none)

======
Previously I installed 1.0.3 in same machine, but I overwrite all files with 1.2.1.. does it cause a trouble?


Can anyone figure out what is the problem?
Thanks very much!

****** ADDITIONAL INFORMATION ******

--- (10 headers 6 lines)---
Using INVITE request as basis request - 79faa8-c0a80203-13c4-3a53f421-bc0c30a-348@192.168.2.3
Sending to 192.168.2.3 : 5060 (non-NAT)
Found peer '852'
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Peer audio RTP is at port 192.168.2.3:2070
Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 8888 in default (domain 192.168.2.1)
list_route: hop: <sip:8888@192.168.2.3:5060>
Transmitting (NAT) to 192.168.2.3:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.3:5060 ;branch=z9hG4bK-3a53f421-bc0c30e-3f06;received=192.168.2.3
From: <sip:elite@192.168.2.3> ;tag=3a53f421-bc0c30c5e1
To: <sip:8888@192.168.2.1:5060>
Call-ID: 79faa8-c0a80203-13c4-3a53f421-bc0c30a-348@192.168.2.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:8888@192.168.2.1>
Content-Length: 0


---
   -- Executing Goto("SIP/852-5b1e", "Yuehfung|s|1") in new stack
   -- Goto (Yuehfung,s,1)
   -- Executing Answer("SIP/852-5b1e", "") in new stack
We're at 192.168.2.1 port 11780
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 192.168.2.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.3:5060 ;branch=z9hG4bK-3a53f421-bc0c30e-3f06;received=192.168.2.3
From: <sip:elite@192.168.2.3> ;tag=3a53f421-bc0c30c5e1
To: <sip:8888@192.168.2.1:5060>;tag=as5c6cdcf9
Call-ID: 79faa8-c0a80203-13c4-3a53f421-bc0c30a-348@192.168.2.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:8888@192.168.2.1>
Content-Type: application/sdp
Content-Length: 158

v=0
o=root 11012 11012 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 11780 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
Comments:By: Serge Vecher (serge-v) 2005-12-14 09:22:40.000-0600

Just to be sure the old binaries are gone, do 'make clean' prior to 'make install'. If the problem persists, at Asterisk prompt do 'set verbose 4', 'set debug 4', enable sip debug and post a complete log here, not just a snippet -- as an attachment. Log should never be posted inline, unless a couple of lines.
By: Jason Chan, Hong Kong (jason_polaris) 2005-12-14 10:06:22.000-0600

Sorry.. i am pretty newbie to asterisk..
how to write the log to a file? I can found /var/log/asterisk/messages
but there is no sip information whatever I type "sip debug" in CLI....
but there is a lot of SIP debug info on my screen...
By: twisted (twisted) 2005-12-14 10:18:59.000-0600

The bugtracker is not a technical support forum.  Please go to either the mailing lists (lists.digium.com) specifically, asterisk-users, or IRC (irc.freenode.net, #asterisk) for end-user support.