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CW_ASN

2003-Sep-25 13:01 UTC

[Asterisk-Users] SIP codecs Errors

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@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@172.16.254.96>
To: <sip:2060@172.16.254.96;user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@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@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@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@172.16.254.96>
To: <sip:2060@172.16.254.96;user=phone;phone-context=unknown>
Date: Thu, 25 Sep 2003 16:49:48 ARBUE
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@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@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@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96:5060 SIP/2.0
Via: SIP/2.0/UDP  172.16.254.96:5060
From: "52880472" <sip:52880472@172.16.254.96>
To:
<sip:2060@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@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96>
To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2060@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@172.16.254.96 for seqno 101
(Response)



Anyone knows whats going on?

Regards,


Gus



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Tilghman Lesher

2003-Sep-25 16:22 UTC

head link

[Asterisk-Users] SIP codecs Errors

On Thursday 25 September 2003 15:01, 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
'show codecs' in no way, shape, or form indicates what codecs are
useable in your sip config.  It returns the same output on ALL machines.
Look in your sip.conf for what codecs you have available.

-Tilghman

Mark Spencer

2003-Sep-25 20:08 UTC

head link

[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@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@172.16.254.96>
> To: <sip:2060@172.16.254.96;user=phone;phone-context=unknown>
> Date: Thu, 25 Sep 2003 16:49:48 ARBUE
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@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@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@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@172.16.254.96>
> To: <sip:2060@172.16.254.96;user=phone;phone-context=unknown>
> Date: Thu, 25 Sep 2003 16:49:48 ARBUE
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@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@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@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96:5060 SIP/2.0
> Via: SIP/2.0/UDP  172.16.254.96:5060
> From: "52880472" <sip:52880472@172.16.254.96>
> To:
<sip:2060@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@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96>
> To:
<sip:2060@172.16.254.96;user=phone;phone-context=unknown>;tag=as2767183f
> Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2060@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@172.16.254.96 for
seqno 101 (Response)
>
>
>
> Anyone knows whats going on?
>
> Regards,
>
>
> Gus
>
>
>
>

mattf

2003-Sep-26 08:59 UTC

head link

[Asterisk-Users] SIP codecs Errors

Me too. Please fix this soon please somebody.

MATT---

-----Original Message-----
From: Bartosz Jozwiak [mailto:bartek@cq-link.sr]
Sent: Friday, September 26, 2003 7:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] SIP codecs Errors



I think there is a problem in CVS because yesterday I updated Asterisk from
CVS and I had the same problem with codecs. When I went back with CVS
everything was working again normal.

-- Bart

----- Original Message ----- 
From: "Tilghman Lesher" <tilghman@mail.jeffandtilghman.com>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, September 25, 2003 8:22 PM
Subject: Re: [Asterisk-Users] SIP codecs Errors

> On Thursday 25 September 2003 15:01, 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
>
> 'show codecs' in no way, shape, or form indicates what codecs are
> useable in your sip config.  It returns the same output on ALL machines.
> Look in your sip.conf for what codecs you have available.
>
> -Tilghman
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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