[AsteriskBrasil] Ligaões de Entrada

Eduardo - Ustel eduardo em ustel.com.br
Terça Setembro 29 10:27:53 BRT 2009


Verifque a configuração da central pelo que me parece ela esta mandando 2650 para o asterisk e o mesmo esta aguardando 1000, isso ocorre quando a central esta programada para mandar o MCDU e como o asterisk não encontrou a extensão 2650 no contexto encerra a chamada

From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=20934ac1daa21-208
To: <sip:+1000 em 10.201.201.1:5060>

  ----- Original Message ----- 
  From: Flavio Miranda 
  To: Asterisk 
  Sent: Tuesday, September 29, 2009 10:04 AM
  Subject: [AsteriskBrasil] Ligaões de Entrada


   
    Caro Felipe,
   
   Como solicitado em e-mail anterior, segue o Degug.
   
   
   
  Connected to Asterisk 1.4.26.2 currently running on emax (pid = 5304)

  ####Mensagem gerada quando ligo para o ramal 1000 no Asterisk
  [Sep 29 10:56:56] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel

   
  #####DEBUG##################
  emax*CLI> sip set debug ip 10.201.201.250
  SIP Debugging Enabled for IP: 10.201.201.250
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  INVITE sip:+1000 em 10.201.201.1:5060 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=20934ac1daa21-208
  To: <sip:+1000 em 10.201.201.1:5060>
  Call-ID: 2ff-4ac1daa2-0-208 em 10.201.201.250
  CSeq: 1 INVITE
  Contact: <sip:+2650 em 10.201.201.250:5060>
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Max-Forwards: 70
  Session-Expires: 1800
  Supported: timer,100rel
  Require: 100rel
  Privacy: none
  P-Asserted-Identity: "+2650" <sip:+2650 em 10.201.201.1:5060>
  Content-Type: application/sdp
  Content-Length:   181
  v=0
  o=IPS 23502 0 IN IP4 10.201.201.250
  s=IPS
  c=IN IP4 10.201.201.250
  t=0 0
  m=audio 10050 RTP/AVP 0 101
  a=rtpmap:0 PCMU/8000/1
  a=ptime:20
  a=rtpmap:101 telephone-event/8000
  <------------->
  --- (17 headers 9 lines) ---
  <--- Transmitting (no NAT) to 10.201.201.250:5060 --->
  SIP/2.0 420 Bad extension (unsupported)
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208;received=10.201.201.250
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=20934ac1daa21-208
  To: <sip:+1000 em 10.201.201.1:5060>;tag=as4153cf25
  Call-ID: 2ff-4ac1daa2-0-208 em 10.201.201.250
  CSeq: 1 INVITE
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Date: Tue, 29 Sep 2009 13:57:03 GMT
  Unsupported: 100rel
  Content-Length: 0

  <------------>
  [Sep 29 10:57:03] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel
  Scheduling destruction of SIP dialog '2ff-4ac1daa2-0-208 em 10.201.201.250' in 32000 ms (Method: INVITE)
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  ACK sip:+1000 em 10.201.201.1:5060 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=20934ac1daa21-208
  To: <sip:+1000 em 10.201.201.1:5060>;tag=as4153cf25
  Call-ID: 2ff-4ac1daa2-0-208 em 10.201.201.250
  CSeq: 1 ACK
  user-agent: IPS
  Max-Forwards: 70
  Content-Length: 0

  <------------->
  --- (9 headers 0 lines) ---
  Really destroying SIP dialog '2ff-4ac1daa2-0-208 em 10.201.201.250' Method: ACK
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4f8f17f8;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7754e9ac
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 3353e7306217b72d0a0f992d6e530f39 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:04 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4f8f17f8;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7754e9ac
  To: <sip:10.201.201.250>
  Call-ID: 3353e7306217b72d0a0f992d6e530f39 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '3353e7306217b72d0a0f992d6e530f39 em 10.201.201.1' Method: OPTIONS
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  INVITE sip:+1000 em 10.201.201.1:5060 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=4c474ac1daac1-210
  To: <sip:+1000 em 10.201.201.1:5060>
  Call-ID: 7d3e-4ac1daac-0-210 em 10.201.201.250
  CSeq: 1 INVITE
  Contact: <sip:+2650 em 10.201.201.250:5060>
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Max-Forwards: 70
  Session-Expires: 1800
  Supported: timer,100rel
  Require: 100rel
  Privacy: none
  P-Asserted-Identity: "+2650" <sip:+2650 em 10.201.201.1:5060>
  Content-Type: application/sdp
  Content-Length:   181
  v=0
  o=IPS 13436 0 IN IP4 10.201.201.250
  s=IPS
  c=IN IP4 10.201.201.250
  t=0 0
  m=audio 10060 RTP/AVP 0 101
  a=rtpmap:0 PCMU/8000/1
  a=ptime:20
  a=rtpmap:101 telephone-event/8000
  <------------->
  --- (17 headers 9 lines) ---
  <--- Transmitting (no NAT) to 10.201.201.250:5060 --->
  SIP/2.0 420 Bad extension (unsupported)
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210;received=10.201.201.250
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=4c474ac1daac1-210
  To: <sip:+1000 em 10.201.201.1:5060>;tag=as3b199582
  Call-ID: 7d3e-4ac1daac-0-210 em 10.201.201.250
  CSeq: 1 INVITE
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Date: Tue, 29 Sep 2009 13:57:12 GMT
  Unsupported: 100rel
  Content-Length: 0

  <------------>
  [Sep 29 10:57:12] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel
  Scheduling destruction of SIP dialog '7d3e-4ac1daac-0-210 em 10.201.201.250' in 32000 ms (Method: INVITE)
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  ACK sip:+1000 em 10.201.201.1:5060 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210
  From: "+2650" <sip:+2650 em 10.201.201.1:5060>;tag=4c474ac1daac1-210
  To: <sip:+1000 em 10.201.201.1:5060>;tag=as3b199582
  Call-ID: 7d3e-4ac1daac-0-210 em 10.201.201.250
  CSeq: 1 ACK
  user-agent: IPS
  Max-Forwards: 70
  Content-Length: 0

  <------------->
  --- (9 headers 0 lines) ---
  Really destroying SIP dialog '7d3e-4ac1daac-0-210 em 10.201.201.250' Method: ACK
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK6b8cee75;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7c556238
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 04fb1766622cc1855bbf67905f6c3c88 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:13 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK6b8cee75;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7c556238
  To: <sip:10.201.201.250>
  Call-ID: 04fb1766622cc1855bbf67905f6c3c88 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '04fb1766622cc1855bbf67905f6c3c88 em 10.201.201.1' Method: OPTIONS
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK5e2e8d2f;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as3429ba79
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 22c4ca806944474364ca70753048a417 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:13 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK3a337600;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as18f7ee3d
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 626ceab217965f8b39c3ae212a74ca25 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:14 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK5e2e8d2f;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as3429ba79
  To: <sip:10.201.201.250>
  Call-ID: 22c4ca806944474364ca70753048a417 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '22c4ca806944474364ca70753048a417 em 10.201.201.1' Method: OPTIONS
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK3a337600;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as18f7ee3d
  To: <sip:10.201.201.250>
  Call-ID: 626ceab217965f8b39c3ae212a74ca25 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '626ceab217965f8b39c3ae212a74ca25 em 10.201.201.1' Method: OPTIONS
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK66ad830a;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7eecb8c9
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 2dedd7755f6bd5eb19473f6602150428 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:14 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK66ad830a;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7eecb8c9
  To: <sip:10.201.201.250>
  Call-ID: 2dedd7755f6bd5eb19473f6602150428 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '2dedd7755f6bd5eb19473f6602150428 em 10.201.201.1' Method: OPTIONS
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:1000 em 10.201.201.6:56702;rinstance=09d7015650b931f8 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK7ac0c8f5;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as0d2c4b3a
  To: <sip:1000 em 10.201.201.6:56702;rinstance=09d7015650b931f8>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 59d07ee263e0180f5233274833d14d1d em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:14 GMT
  llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> 
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK7ac0c8f5;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as0d2c4b3a
  To: <sip:1000 em 10.201.201.6:56702;rinstance=09d7015650b931f8>
  Call-ID: 59d07ee263e0180f5233274833d14d1d em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '59d07ee263e0180f5233274833d14d1d em 10.201.201.1' Method: OPTIONS
  Reliably Transmitting (no NAT) to 10.201.201.250:5060:
  OPTIONS sip:10.201.201.250 SIP/2.0
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4b9d60e2;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7becd38c
  To: <sip:10.201.201.250>
  Contact: <sip:asterisk em 10.201.201.1>
  Call-ID: 31bc7fe044f8ea6b6312f10a5f685342 em 10.201.201.1
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Tue, 29 Sep 2009 13:57:16 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Content-Length: 0

  ---
  emax*CLI> sip set debug ip 10.201.201.250
  <--- SIP read from 10.201.201.250:5060 --->
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4b9d60e2;rport
  From: "asterisk" <sip:asterisk em 10.201.201.1>;tag=as7becd38c
  To: <sip:10.201.201.250>
  Call-ID: 31bc7fe044f8ea6b6312f10a5f685342 em 10.201.201.1
  CSeq: 102 OPTIONS
  Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
  user-agent: IPS
  Supported: timer,100rel
  Content-Length: 0

  <------------->
  --- (10 headers 0 lines) ---
  Really destroying SIP dialog '31bc7fe044f8ea6b6312f10a5f685342 em 10.201.201.1' Method: OPTIONS
  emax*CLI> sip set debug off
  SIP Debugging Disabled
  emax*CLI> 


  Att,
   
  Flavio Roberto Miranda
  MSN:flaviormiranda em hotmail.com
  Skype: flaviormiranda




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  Esta mensagem foi verificada pelo NOD32 sistema antivírus
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