[Asterisk-Users] SIP codecs Errors
CW_ASN
cw_asn at fibertel.com.ar
Fri Sep 26 04:33:25 MST 2003
Mark:
When I update from CVS and delete "allow=all" in sip.conf it works great. If
"allow=all" remains in sip.conf it doesn't work.
Anyway, It's working again.
Thanks a lot!
Regards,
Gus
----- Original Message -----
From: "Mark Spencer" <markster at digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, September 26, 2003 12:08 AM
Subject: Re: [Asterisk-Users] SIP codecs Errors
> 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
> >
> >
> >
> >
>
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