[Asterisk-Users] SIP codecs Errors

CW_ASN Thu, 25 Sep 2003 13:12:23 -0700

Hi all:

I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message:

*CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs!

The "show codecs" command shows:

*CLI> show codecs
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 << 4) MPEG-2 layer 3
32 (1 << 5) ADPCM
64 (1 << 6) 16 bit Signed Linear PCM
128 (1 << 7) LPC10
256 (1 << 8) G.729A audio
512 (1 << 9) SpeeX
1024 (1 << 10) iLBC
65536 (1 << 16) JPEG image
131072 (1 << 17) PNG image
262144 (1 << 18) H.261 Video
524288 (1 << 19) H.263 Video

The "sip debug" show the following:

*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:[EMAIL PROTECTED]>
To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: [EMAIL PROTECTED]
Cisco-Guid: 1091135146-4006089175-2409868731-3383986922
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1064519388
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 167

v=0
o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96
s=SIP Call
c=IN IP4 172.16.254.96
t=0 0
m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535

15 headers, 6 lines
Using latest request as basis request
Sending to 172.16.254.96 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format UNKN
Capabilities: us - 0, them - 269/0, combined - 0
Non-codec capabilities: us - 1, them - 0, combined - 0
WARNING[1125329600]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs!
Sip read:
INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:[EMAIL PROTECTED]>
To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: [EMAIL PROTECTED]
Cisco-Guid: 1091135146-4006089175-2409868731-3383986922
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1064519388
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 167

v=0
o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96
s=SIP Call
c=IN IP4 172.16.254.96
t=0 0
m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535

15 headers, 6 lines
Ignoring this request
Looking for 2060 in default
list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:[EMAIL PROTECTED]>
To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0

to 172.16.254.96:5060
-- Executing VoiceMail("SIP/-0812ba78", "u2060") in new stack
We're at 172.16.254.96 port 16464
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:[EMAIL PROTECTED]>
To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 109