[Asterisk-Users] SIP codecs Errors
CW_ASN
cw_asn at fibertel.com.ar
Thu Sep 25 13:01:49 MST 2003
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:2060 at 172.16.254.96;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
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:52880472 at 172.16.254.96:5060;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:2060 at 172.16.254.96;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
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:52880472 at 172.16.254.96:5060;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:52880472 at 172.16.254.96:5060;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
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:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm-theperson'
Sip read:
BYE sip:2060 at 172.16.254.96:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 1064519388
CSeq: 102 BYE
Content-Length: 0
11 headers, 0 lines
Sending to 172.16.254.96 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Length: 0
to 172.16.254.96:5060
== Spawn extension (default, 2060, 1) exited non-zero on 'SIP/-0812ba78'
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.96:5060
From: "52880472" <sip:52880472 at 172.16.254.96>
To: <sip:2060 at 172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060 at 172.16.254.96>
Content-Type: application/sdp
Content-Length: 109
v=0
o=root 3781 3781 IN IP4 172.16.254.96
s=session
c=IN IP4 172.16.254.96
t=0 0
m=audio 16464 RTP/AVP
to 172.16.254.96:5060
WARNING[1125329600]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries exceeded on call 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA at 172.16.254.96 for seqno 101 (Response)
Anyone knows whats going on?
Regards,
Gus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0fc82848/attachment.htm
More information about the asterisk-users
mailing list