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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>É parece ser um problema entre a conexão do SipS ao asterisk.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>O asterisk depois de alguns segundos falando diz que não esta ok
a conexão e o SipS derruba a chamada.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Já descartei ser codecs, pois já tentei de tudo.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Não sei mais o que fazer.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Fiz testes com outro asterisk na mesma versão e mesmo problema.
Porem no terceiro asterisk que testei de mesma versão tudo funciona. Comparei
os logs e vi que configurações e são as mesmas.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>De asterisk para asterisk trunk sip funciona perfeito.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asteriskbrasil-bounces@listas.asteriskbrasil.org
[mailto:asteriskbrasil-bounces@listas.asteriskbrasil.org] <b>Em nome de </b>Felippe<br>
<b>Enviada em:</b> quarta-feira, 27 de outubro de 2010 16:34<br>
<b>Para:</b> asteriskbrasil@listas.asteriskbrasil.org<br>
<b>Assunto:</b> [AsteriskBrasil] Mensagem Maximum retries exceeded on
transmission após hangup na chamada<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Saudações.<br>
<br>
Veja se alguem consegue me ajudar na solucao deste problema:<br>
<br>
Estou tendo um problema quanto ao recebimento de chamadas no
meu asterisk no qual não consigo identificar de onde pode esta vindo tal erro.
Já perdi 3 dias tentando de tudo mas ainda nada deu certo. As chamadas são
encaminhadas de um OpenSips ate ele. Ate 5 dias atras vinha funcionando a
mais de 2 anos. Porem esse final de semana tive que trocar o servidor asterisk
no qual recebe as chamdas e reinstalei tudo novamente. Mesma versao do
asterisk, etc. Porem começou esses problema.<br>
Tenho uma URA de entrada, entao o cliente escuta a opcao de 1 a 10 por exemplo.
Ele liga no meu DID ou liga do Opensips que encaminha para meu asterisk, ouve a
URA e na quarta ou quinta opcao fica mudo e tenho a mensagem na cli:<br>
Não estou atras de NAT, o asterisk faz a conexao pppoe. <br>
Versao 1.4.21.2<br>
<br>
[Oct 27 16:03:01] WARNING[12859]: chan_sip.c:1950 retrans _pkt: Maximum retries
exceeded on transmission <a
href="mailto:41D42534-E12B11DF-B9A2BFFB-47F79513@200.200.200.200">41D42534-E12B11DF-B9A2BFFB-47F79513@200.200.200.200</a>
for seqno 101 (Critical Response)<br>
[Oct 27 16:03:01] WARNING[12859]: chan_sip.c:1972 retrans_pkt: Hanging up call <a
href="mailto:41D42534-E12B11DF-B9A2BFFB-47F79513@200.200.200.200">41D42534-E12B11DF-B9A2BFFB-47F79513@200.200.200.200</a>
- no reply to our critical packet.<br>
== Spawn extension (ura_comeco, s, 1) exited non-zero on
'SIP/provedor-08d3c9b8'<br>
<br>
<br>
Nao sei se posso enviar o debug por aqui mas ja enviando.. tenho isso
em todo inicio ao final:<br>
<br>
<br>
t 27 16:26:11] DEBUG[13421]: pbx.c:1842 pbx_extension_helper: Launching 'Wait'<br>
-- Executing [s@ura:5]
Wait("SIP/provedor-08a7cbc8", "1") in new stack<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:287 do_state_change: Changing
state for SIP/provedor-08a7cbc8 - state 4 (Invalid)<br>
[Oct 27 16:26:11] D EBUG[13365]: devicestate.c:161 ast_device_state: No
provider found, checking channel drivers for SIP - provedor<br>
[Oct 27 16:26:11] DEBUG[13365]: chan_sip.c:16004 sip_devicestate: Checking
device state for peer provedor<br>
[Oct 27 16:26:11] DEBUG[13382]: app_queue.c:659 handle_statechange: Device
'SIP/provedor-08a7cbc8' changed to state '4' (Invalid) but we don't care
because they're not a member of any queue.<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:287 do_state_change: Changing
state for SIP/provedor- state 4 (Invalid)<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:161 ast_device_state: No provider
found, checking channel drivers for SIP - provedor-08a7cbc8<br>
[Oct 27 16:26:11] DEBUG[13365]: chan_sip.c:16004 sip_devicestate: Checking
device state for peer provedor-08a7cbc8<br>
[Oct 27 16:26:11] DEBUG[13382]: app_queue.c:659 handle_statechange: Device
'SIP/provedor changed to state '4' (Invalid) but we don't care because they're
not a member of any que ue.<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:287 do_state_change: Changing
state for SIP/provedor-08a7cbc8 - state 4 (Invalid)<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:161 ast_device_state: No provider
found, checking channel drivers for SIP - provedor<br>
[Oct 27 16:26:11] DEBUG[13365]: chan_sip.c:16004 sip_devicestate: Checking
device state for peer provedor<br>
[Oct 27 16:26:11] DEBUG[13382]: app_queue.c:659 handle_statechange: Device
'SIP/provedor-08a7cbc8' changed to state '4' (Invalid) but we don't care
because they're not a member of any queue.<br>
[Oct 27 16:26:11] DEBUG[13365]: devicestate.c:287 do_state_change: Changing
state for SIP/provedor- state 4 (Invalid)<br>
[Oct 27 16:26:11] DEBUG[13382]: app_queue.c:659 handle_statechange: Device
'SIP/provedor changed to state '4' (Invalid) but we don't care because they're
not a member of any queue.<br>
[Oct 27 16:26:12] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retran smission 2 to 200 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:12] DEBUG[13367]: sched.c:204 sched_settime: Request to schedule
in the past?!?!<br>
[Oct 27 16:26:12] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:12] DEBUG[13421]: pbx.c:1842 pbx_extension_helper: Launching
'GotoIfTime'<br>
-- Executing [s@ura:6]
GotoIfTime("SIP/provedor-08a7cbc8",
"7:55-19:30|mon-fri|*|*?ura_comeco|s|1(inicio)") in new stack<br>
-- Goto (ura_comeco,s,1)<br>
[Oct 27 16:26:12] DEBUG[13421]: pbx.c:1842 pbx_extension_helper: Launching
'BackGround'<br>
-- Executing [s@ura_comeco:1]
BackGround("SIP/provedor-08a7cbc8", "ura_inicio") in new
stack<br>
[Oct 27 16:26:12] DEBUG[13421]: channel.c:2808 set_format: Set channel
SIP/provedor-08a7cbc8 to write format slin<br>
[Oct 27 16:26:12] DEBUG[13421]: rtp.c:2769 ast_rtp_write: Ooh, format changed from
unknown to g729<br>
[Oct 27 16:26:12] DEBUG[13421]: rtp.c:2786 ast_rtp_write: Created smoother:
format: 256 ms: 20 len: 20<br>
[Oct 27 16:26:12] DEBUG[13421]: channel.c:1804 ast_settimeout: Scheduling timer
at 160 sample intervals<br>
-- <SIP/provedor-08a7cbc8> Playing 'ura_inicio'
(language 'pt_BR')<br>
[Oct 27 16:26:13] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:13] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:15] DEBUG[13421]: rtp.c:879 ast_rtcp_read: Got RTCP report of 136
bytes<br>
[Oct 27 16:26:15] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:16] DEBUG[13421]: rtp.c:879 ast_rtcp_read: Got RTCP report of 136
bytes<br>
[Oct 27 16:26:18] DEBUG[13367]: chan_sip.c:1928 retrans_pkt: ** SIP timers:
Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #143))<br>
[Oct 27 16:26:22] WARNING[13367]: chan_sip.c:1950 retrans_pkt: Maximum retries
exceeded on transmission <a
href="mailto:85973717-E12E11DF-BF07BFFB-47F79513@200.196.28.101">85973717-E12E11DF-BF07BFFB-47F79513@200.200.200.200<span
style='color:black'> </span></a> for seqno 101 (Critical Response)<br>
[Oct 27 16:26:22] DEBUG[13367]: chan_sip.c:1657 sip_alreadygone: Setting
SIP_ALREADYGONE on dialog <a
href="mailto:85973717-E12E11DF-BF07BFFB-47F79513@200.200.200.200">85973717-E12E11DF-BF07BFFB-47F79513@200.200.200.200</a>
[Oct 27 16:26:22] WARNING[13367]: chan_sip.c:1972 retrans_pkt: Hanging up call <a
href="mailto:85973717-E12E11DF-BF07BFFB-47F79513@200.196.28.101">85973717-E12E11DF-BF07BFFB-47F79513@200.200.200.200<span
style='color:black'> </span></a> - no reply to our critical packet.<br>
[Oct 27 16:26:22] DEBUG[13421]: channel.c:1804 ast_settimeout: Scheduling timer
at 0 sample intervals<br>
[Oct 27 16:26:22] DEBUG[13421]: channel.c:2808 set_format: Set channel
SIP/provedor-08a7cbc8 to write format g729<br>
[Oct 27 16:26:22] DEBUG[13421]: pbx.c:2438 __ast_pbx_run: Spawn extension
(ura_comeco,s,1) exited non-zero on 'SIP/provedor-08a7cbc8'<br>
== Spawn extension (ura_comeco, s, 1) exited non-zero on
'SIP/provedor-08a7cbc8'<br>
[Oct 27 16:26:22] DEBUG[13421]: channel.c:1384 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/provedor-08a7cbc8'<br>
[Oct 27 16:26:22] DEBUG[13421]: channel.c:1483 ast_hangup: Hanging up channel
'SIP/provedor-08a7cbc8'<br>
[Oct 27 16:26:22] DEBUG[13421]: chan_sip.c:3522 sip_hangup: Hangup call
SIP/provedor-08a7cbc8, SIP callid 85973717-E12E11DF-BF07BFFB-47F79513@200.200.200.200<span
style='color:black'> </span>)<br>
[Oct 27 16:26:22] DEBUG[13421]: chan_sip.c:3210 update_call_counter: Updating
call counter for incoming call<br>
<br>
<br>
Cordial,<br>
<br>
Felippe <o:p></o:p></p>
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