|
I tried to change the codec to ulaw but still
cannot do anything. I got this on my Asterisk box: --------------------------------------------------------------------------------- Found RTP audio format 0 Peer audio RTP is at port
192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer -
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1
(telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 180 Ringing Call-ID:
[EMAIL PROTECTED] Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From:
"1656222"<sip:[EMAIL PROTECTED]>;tag=as20454a28 To:
<sip:[EMAIL PROTECTED]>;tag=c0a800fe-b User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP
192.168.0.1:5060;branch=z9hG4bK63e689bf;rport v=0 o=Quintum 3 3131 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 0 c=IN IP4 192.168.0.254 a=rtpmap:0 pcmu/8000/1 --- (9 headers 8 lines)--- Found RTP audio format 0 Peer audio RTP is at port
192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer -
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1
(telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) And this on the quintum box : --------------------------------------------------------------------------------- CH :
29263617:sip[0]: sip:RcvIncomingCall CH :
29263618:sip[0]: osipcall:RcvSetup, my media type=4 CH :
29263618:chsip : bandwidth info: max=-1 cur=12600. CH :
29263618:chsip: Media present in Setup CH :
29263618:chsip: Setting remote rtp port=192.168.0.1:19126. CH :
29263618:Remote side packet saver version = 2. CH :
29263618:CallInfo[0xd3c82c]: origCalled.digit(16562227279561) . CH :
29263618:sip[34/0]: osipcall:stackSendCallProc CH : 29263618:sent
message to sip: msg=7; ua=1 CH :
29263618:Routing requested for: public(1) orig=16562227279561 public(1)
normalized=16562227279561 route code= tg=0. CH : 29263618:1
match(es) found: 3 CH :
29263618:CasTG[3]: newTermCall: selected line=256 chan=256. CH : 29263618:Route
response(34): result=1 cause=0. CH :
29263618:udp connect: 9 11 CH :
29263618:
c0a800fe 10240 c0a80001 19126 CH :
29263618:TBCSM[34]: Setup from peer=0xd3c808 NP=0x0 NT=0x0. CH :
29263618:OrigNum=16562227279561 NormNum=16562227279561 TranNum=0227279561
OrigDest=. CH : 29263618:[2:
1] sent message to cas: Setup CH :
29263630:tsi connect: 001 202 01 CH :
29263630:TsiConnXlate: 0:1, 2:2 CH : 29263657:tsi
disconnect: 001 202 01 CH : 29263657:TsiDiscXlate:
0:1, 2:2 CH : 29263657:[2:
1] received message from cas: Call-Proc CH :
29263664:tsi connect: 001 210 10 CH :
29263664:TsiConnXlate: 2:10, 0:1 CH : 29263884:tsi
disconnect: 001 210 10 CH :
29263884:TsiDiscXlate: 2:10, 0:1 CH : 29263884:[2:
1] received message from cas: Alert CH :
29263884:sip[34/0]: osipcall:stackSendProg CH : 29263884:sent
message to sip: msg=9; ua=1 CH :
29263884:tsi connect: 001 209 01 CH :
29263884:TsiConnXlate: 0:1, 2:9 CH : 29263884:tsi
connect: 001 209 10 CH :
29263884:TsiConnXlate: 2:9, 0:1 CH :
29263884:sip[34/0]: osipcall:stackSendAlert CH : 29263884:sent
message to sip: msg=10; ua=1 Followed are my quintum dsp settings: ----------------------------------------------------------------- Voice Coding algorithm = 9 à for G711 U-law=9 Voice Information Field size = 1280 bits Silence Suppression = Enable(1) Minimum Jitter buffer = 60 msec Maximum Jitter buffer = 150 msec Receive Gain (PCM -> IP) = 2 dB Transmit Gain (IP -> PCM) = 0 dB Digit Relay = 0 Fax Relay Type = 0 Fax Maximum Rate = 144 Fax Playout FIFO nominal delay = 600 Fax Coding = 0 Packet Saver = Disabled Idle Time = 0 Answer Supervision Options = 0 From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Neill Wilkinson In the quintum also check
you have a codec profile: Example (below has alaw and
G729 configure in the codec profile): CodecProfile-default :
name
: (Not Set)
name
VoiceCodecAttached[1] : VoiceCodec-1
VoiceCodecAttached[2] : VoiceCodec-2
VoiceCodecAttached[3..8] : (unspecified) CodecProfile-default :
name
: (Not Set)
name
VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2]
: VoiceCodec-2
VoiceCodecAttached[3..8] : (unspecified) config-VoiceCodec-2* show VoiceCodec-2 :
name
: (Not Set)
name
CodecVoiceCoding
:
8
G.711 A-Law
CodecPayloadSize
:
1280
bits
config-VoiceCodec-2* this profile should be
attached to you IP Routing Group (IPRG). |
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
