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<P><FONT face="Courier New" size=2>Hi all: </FONT></P>
<P><FONT face="Courier New" size=2>I recently update a system from CVS (Asterisk
CVS-09/25/03-15:58:42), and I receiving the following message: </FONT></P>
<P><FONT face="Courier New" size=2>*CLI> WARNING[1187305408]: File
chan_sip.c, Line 1864 (process_sdp): No compatible codecs!</FONT></P>
<P><FONT face="Courier New" size=2>The "show codecs" command shows: </FONT></P>
<P><FONT face="Courier New" size=2>*CLI> show codecs<BR>1 (1 << 0)
G.723.1<BR>2 (1 << 1) GSM<BR>4 (1 << 2) G.711 u-law<BR>8 (1 <<
3) G.711 A-law<BR>16 (1 << 4) MPEG-2 layer 3<BR>32 (1 << 5)
ADPCM<BR>64 (1 << 6) 16 bit Signed Linear PCM<BR>128 (1 << 7)
LPC10<BR>256 (1 << 8) G.729A audio<BR>512 (1 << 9) SpeeX<BR>1024 (1
<< 10) iLBC<BR>65536 (1 << 16) JPEG image<BR>131072 (1 << 17)
PNG image<BR>262144 (1 << 18) H.261 Video<BR>524288 (1 << 19) H.263
Video</FONT></P>
<P><FONT face="Courier New" size=2>The "sip debug" show the
following:</FONT></P>
<P><FONT face="Courier New" size=2>*CLI> sip debug<BR>SIP Debugging
Enabled<BR>Sip read:<BR>INVITE
sip:2060@172.16.254.96;user=phone;phone-context=unknown SIP/2.0<BR>Via:
SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown><BR>Date: Thu, 25
Sep 2003 16:49:48 ARBUE<BR>Call-ID: <A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>Cisco-Guid:
1091135146-4006089175-2409868731-3383986922<BR>User-Agent: Cisco VoIP Gateway/
IOS 12.x/ SIP enabled<BR>CSeq: 101 INVITE<BR>Max-Forwards: 6<BR>Timestamp:
1064519388<BR>Contact:
<sip:52880472@172.16.254.96:5060;user=phone><BR>Expires:
180<BR>Content-Type: application/sdp<BR>Content-Length: 167</FONT></P>
<P><FONT face="Courier New" size=2>v=0<BR>o=CiscoSystemsSIP-GW-UserAgent 8010
6925 IN IP4 172.16.254.96<BR>s=SIP Call<BR>c=IN IP4 172.16.254.96<BR>t=0
0<BR>m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535</FONT></P>
<P><FONT face="Courier New" size=2>15 headers, 6 lines<BR>Using latest request
as basis request<BR>Sending to 172.16.254.96 : 5060 (non-NAT)<BR>Found audio
format ALAW<BR>Found audio format UNKN<BR>Found audio format UNKN<BR>Found audio
format UNKN<BR>Found audio format UNKN<BR>Found audio format UNKN<BR>Found audio
format ULAW<BR>Found audio format UNKN<BR>Capabilities: us - 0, them - 269/0,
combined - 0<BR>Non-codec capabilities: us - 1, them - 0, combined -
0<BR>WARNING[1125329600]: File chan_sip.c, Line 1864 (process_sdp): No
compatible codecs!<BR>Sip read:<BR>INVITE
sip:2060@172.16.254.96;user=phone;phone-context=unknown SIP/2.0<BR>Via:
SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown><BR>Date: Thu, 25
Sep 2003 16:49:48 ARBUE<BR>Call-ID: <A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>Cisco-Guid:
1091135146-4006089175-2409868731-3383986922<BR>User-Agent: Cisco VoIP Gateway/
IOS 12.x/ SIP enabled<BR>CSeq: 101 INVITE<BR>Max-Forwards: 6<BR>Timestamp:
1064519388<BR>Contact:
<sip:52880472@172.16.254.96:5060;user=phone><BR>Expires:
180<BR>Content-Type: application/sdp<BR>Content-Length: 167</FONT></P>
<P><FONT face="Courier New" size=2>v=0<BR>o=CiscoSystemsSIP-GW-UserAgent 8010
6925 IN IP4 172.16.254.96<BR>s=SIP Call<BR>c=IN IP4 172.16.254.96<BR>t=0
0<BR>m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535</FONT></P>
<P><FONT face="Courier New" size=2>15 headers, 6 lines<BR>Ignoring this
request<BR>Looking for 2060 in default<BR>list_route: hop:
<sip:52880472@172.16.254.96:5060;user=phone><BR>Transmitting (no
NAT):<BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
172.16.254.96:5060<BR>From: "52880472" <sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Length:
0</FONT></P><FONT face="Courier New" size=2>
<P><BR> to 172.16.254.96:5060<BR> -- Executing
VoiceMail("SIP/-0812ba78", "u2060") in new stack<BR>We're at 172.16.254.96 port
16464<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR> == Parsing
'/etc/asterisk/voicemail.conf': Found<BR> -- Playing
'vm-theperson'<BR>Sip read:<BR>BYE sip:2060@172.16.254.96:5060 SIP/2.0<BR>Via:
SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Date:
Thu, 25 Sep 2003 16:49:48 ARBUE<BR>Call-ID: <A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>User-Agent:
Cisco VoIP Gateway/ IOS 12.x/ SIP enabled<BR>Max-Forwards: 6<BR>Timestamp:
1064519388<BR>CSeq: 102 BYE<BR>Content-Length: 0</P>
<P><BR>11 headers, 0 lines<BR>Sending to 172.16.254.96 : 5060
(non-NAT)<BR>Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
172.16.254.96:5060<BR>From: "52880472" <sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
102 BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Length: 0</P>
<P><BR> to 172.16.254.96:5060<BR> == Spawn extension (default, 2060,
1) exited non-zero on 'SIP/-0812ba78'<BR>Retransmitting #1 (no NAT):<BR>SIP/2.0
200 OK<BR>Via: SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR>Retransmitting #2 (no NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR>Retransmitting #3 (no NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR>Retransmitting #4 (no NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR>Retransmitting #5 (no NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP 172.16.254.96:5060<BR>From: "52880472"
<sip:52880472@172.16.254.96><BR>To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f<BR>Call-ID:
<A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:2060@172.16.254.96><BR>Content-Type:
application/sdp<BR>Content-Length: 109</P>
<P>v=0<BR>o=root 3781 3781 IN IP4 172.16.254.96<BR>s=session<BR>c=IN IP4
172.16.254.96<BR>t=0 0<BR>m=audio 16464 RTP/AVP</P>
<P> to 172.16.254.96:5060<BR>WARNING[1125329600]: File chan_sip.c, Line 444
(retrans_pkt): Maximum retries exceeded on call <A
href="mailto:410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96">410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96</A>
for seqno 101 (Response)</P>
<P> </P>
<P>Anyone knows whats going on?</P>
<P>Regards,</P>
<P><BR>Gus</P>
<P><FONT face=Arial></FONT></FONT> </P></DIV></FONT></FONT></BODY></HTML>