Faz o teste colocando isso no general do sip.conf:<div>session-timers=originate<br clear="all"><div><div style="text-align:left"><span style="color:rgb(0,0,153);font-family:Verdana,sans-serif;background-color:rgb(255,255,255)"><strong>Fábio Fioresi Paqueli </strong></span></div>
<span style="background-color:rgb(255,255,255)"><span style="font-family:arial,sans-serif">..............................</span><span style="font-family:arial,sans-serif">........................</span></span><div style="text-align:left">
<span style="font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><font color="#999999">NOC - Voxbras</font></span></div><div style="text-align:left"><span style="font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><font color="#999999">(28) 3310-3307</font></span></div>
<div style="text-align:left"><span style="color:rgb(153,153,153);background-color:rgb(255,255,255);font-family:arial,sans-serif;font-size:13px">(28) 9969-9045</span></div><div style="text-align:left"></div><div style="text-align:left">
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<br><br><div class="gmail_quote">On Tue, Aug 13, 2013 at 10:05 AM, Fernando Trilha <span dir="ltr"><<a href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div>Pessoal, sempre que faço uma ligação depois de uns segundos ou cai ou fica mudo...</div><div>Fica assim na console:</div><div><br></div><div>Connected to Asterisk 1.8.23.0 currently running on Asterisk (pid = 16799)</div>
<div>Verbosity was 0 and is now 33</div><div> -- Registered SIP '9962' at <a href="http://177.202.200.238:5060" target="_blank">177.202.200.238:5060</a></div><div> > Saved useragent "Zoiper r18976" for peer 9962</div>
<div>[Aug 13 09:44:35] NOTICE[16819]: chan_sip.c:21575 handle_response_peerpoke: Peer '9962' is now Reachable. (52ms / 2000ms)</div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [04821079198@entrada:1] Dial("SIP/9962-00000000", "SIP/vono/04821079198,90,tTwW") in new stack</div>
<div> == Using SIP RTP CoS mark 5</div><div> -- Called SIP/vono/04821079198</div><div> -- SIP/vono-00000001 is making progress passing it to SIP/9962-00000000</div><div> -- SIP/vono-00000001 answered SIP/9962-00000000</div>
<div>[Aug 13 09:45:39] WARNING[16819]: chan_sip.c:3982 retrans_pkt: Retransmission timeout reached on transmission ODMwYjkxZmM3OWYzZWFhOGUxZWRhOTEzNDYwOTEyMGE. for seqno 2 (Critical Response) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a></div>
<div>Packet timed out after 6400ms with no response</div><div>[Aug 13 09:45:39] WARNING[16819]: chan_sip.c:4011 retrans_pkt: Hanging up call ODMwYjkxZmM3OWYzZWFhOGUxZWRhOTEzNDYwOTEyMGE. - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).</div>
<div> == Spawn extension (entrada, 04821079198, 1) exited non-zero on 'SIP/9962-00000000'</div><div>Asterisk*CLI></div><div><br></div><div>Meu sip está assim...</div><div><br></div><div><div>[general]</div><div>
disallow=all</div><div>allow=ulaw</div><div>port=5070</div><div>bindaddr=0.0.0.0</div><div>context=entrada</div><div>register=><a href="http://fxxxxx@vono.net.br:5060/xxxxx" target="_blank">fxxxxx@vono.net.br:5060/xxxxx</a></div>
<div>
language=pt_BR</div><div>externip=xxx.xxx.xxx.xxx</div><div>localnet=<a href="http://192.168.0.0/24" target="_blank">192.168.0.0/24</a></div><div>localnet=<a href="http://172.16.100.0/255.255.254.0" target="_blank">172.16.100.0/255.255.254.0</a></div>
<div>
externhost=xxx.xxx.xxx.xxx</div><div>externrefresh = 15</div><div>bindport=5070</div><div><br></div><div>[vono] ;o nome do ramal deve ser o mesmo login da vono</div><div>type=friend</div><div>username=xxxxx</div><div>secret=senha</div>
<div>fromuser=xxxxx</div><div>fromdomain=<a href="http://vono.net.br" target="_blank">vono.net.br</a></div><div>context=entrada</div><div>disallow=all</div><div>allow=alaw</div><div>allow=ilbc</div><div>allow=ulaw</div><div>
host=<a href="http://vono.net.br" target="_blank">vono.net.br</a></div>
<div>insecure=port,invite</div><div>qualify=no</div><div>port=5060</div><div>aut=md5</div><div>nat=yes</div><div>canreinvite=no</div><div>dtmfmode=rfc2833</div></div><span class="HOEnZb"><font color="#888888"><div><br></div>
<div><br></div>-- <br><div dir="ltr">Atte.<br>
Fernando Trilha<br>Analista de Suporte <div>f<a href="mailto:ftrilha@gmail.com" target="_blank">trilha@gmail.com</a><br></div><div>::Soluções em informatica e redes corporativas::</div></div>
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Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;<br>
Intercomunicadores para acesso remoto via rede IP. Conheça em <a href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
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ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.<br>
Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.<br>
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