[Asterisk-Users] SIP codecs Errors

Thu Sep 25 20:08:25 MST 2003

Fixed in CVS

On Thu, 25 Sep 2003, CW_ASN wrote:

> 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
>
>
>
>