[Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

[Freddy Setiawan]

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: 2a5bb110693fbc7259ceaf6c3928a050 at 192.168.0.1

Content-Length: 160

Content-Type: application/sdp

CSeq: 102 INVITE

From: "1656222"<sip:1656222 at 192.168.0.1>;tag=as20454a28

To: <sip:16562227279561 at 192.168.0.254>;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       --> 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Neill
Wilkinson
Sent: Monday, June 26, 2006 3:54 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

 

 

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).

 

 

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