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Uma dúvida.<br>
<br>
Seu asterisk está localmente com ip interno e ele tem
redirecionamento do routeador para ip real, certo? (Corrija se eu
estiver correto)<br>
<br>
Ao tentar registrar seu softphone vc usa o ip local ou o ip externo
real?<br>
<br>
Perceba que no siptrace que vc postou, ele não consegue falar com
seu roteador (NAT? Firewall?) : Retransmitting #1 (NAT) to <a
moz-do-not-send="true" href="http://200.193.70.93:2683">200.193.70.93:2683</a>:<br>
<br>
Seus dispositivos estão se registrando e informando que estão com ip
externo: Reg. Contact :
<a class="moz-txt-link-abbreviated" href="mailto:sip:9961@200.193.70.93:24477;rinstance=0f6639093db4a1d7">sip:9961@200.193.70.93:24477;rinstance=0f6639093db4a1d7</a><br>
<br>
Mas ao estabelecer a conexão, usam IP interno no rtp: <br>
<br>
<div><b>o=- 4 2 IN IP4 10.0.0.100</b></div>
<div><b>s=CounterPath X-Lite 3.0</b></div>
<b>
</b>
<div><b>c=IN IP4 10.0.0.100</b><br>
<br>
Seu asterisk consegue falar com o IP <b>10.0.0.100</b> por
exemplo?<br>
</div>
<br>
Esse é um bom caminho de partida, ja deu para perceber que o seu
problema é puramente NAT. Basta apenas fazer as alterações
necessárias para que você consiga "fugir" desse NAT ou então
resolver, para que seu asterisk consiga falar com a rede <b>10.0.0.X</b>
e com a rede <b>192.168.1.X</b><br>
<br>
<div class="moz-cite-prefix">Em 28-10-2013 15:41, Fernando Trilha
escreveu:<br>
</div>
<blockquote
cite="mid:CABbThRCrev+BfsZhaeXFuEcghaPEThtyjEqtnJQNfau1KQgzBw@mail.gmail.com"
type="cite">
<div dir="ltr">Fernando, vamos la...
<div><br>
</div>
<div>
<div>servidor*CLI> sip show settings</div>
<div><br>
</div>
<div><br>
</div>
<div>Global Settings:</div>
<div>----------------</div>
<div> UDP Bindaddress: <a moz-do-not-send="true"
href="http://0.0.0.0:5060">0.0.0.0:5060</a></div>
<div> TCP SIP Bindaddress: Disabled</div>
<div> TLS SIP Bindaddress: Disabled</div>
<div> Videosupport: No</div>
<div> Textsupport: No</div>
<div> Ignore SDP sess. ver.: No</div>
<div> AutoCreate Peer: No</div>
<div> Match Auth Username: No</div>
<div> Allow unknown access: Yes</div>
<div> Allow subscriptions: Yes</div>
<div> Allow overlap dialing: Yes</div>
<div> Allow promisc. redir: No</div>
<div> Enable call counters: No</div>
<div> SIP domain support: No</div>
<div> Realm. auth: No</div>
<div> Our auth realm asterisk</div>
<div> Use domains as realms: No</div>
<div> Call to non-local dom.: Yes</div>
<div> URI user is phone no: No</div>
<div> Always auth rejects: Yes</div>
<div> Direct RTP setup: No</div>
<div> User Agent: Asterisk PBX 1.8.11.0</div>
<div> SDP Session Name: Asterisk PBX 1.8.11.0</div>
<div> SDP Owner Name: root</div>
<div> Reg. context: (not set)</div>
<div> Regexten on Qualify: No</div>
<div> Legacy userfield parse: No</div>
<div> Caller ID: asterisk</div>
<div> From: Domain:</div>
<div> Record SIP history: Off</div>
<div> Call Events: Off</div>
<div> Auth. Failure Events: Off</div>
<div> T.38 support: No</div>
<div> T.38 EC mode: Unknown</div>
<div> T.38 MaxDtgrm: -1</div>
<div> SIP realtime: Disabled</div>
<div> Qualify Freq : 60000 ms</div>
<div> Q.850 Reason header: No</div>
<div> Store SIP_CAUSE: No</div>
<div><br>
</div>
<div>Network QoS Settings:</div>
<div>---------------------------</div>
<div> IP ToS SIP: CS0</div>
<div> IP ToS RTP audio: CS0</div>
<div> IP ToS RTP video: CS0</div>
<div> IP ToS RTP text: CS0</div>
<div> 802.1p CoS SIP: 4</div>
<div> 802.1p CoS RTP audio: 5</div>
<div> 802.1p CoS RTP video: 6</div>
<div> 802.1p CoS RTP text: 5</div>
<div> Jitterbuffer enabled: No</div>
<div><br>
</div>
<div>Network Settings:</div>
<div>---------------------------</div>
<div> SIP address remapping: Disabled, no localnet list</div>
<div> Externhost: <none></div>
<div> Externaddr: (null)</div>
<div> Externrefresh: 10</div>
<div><br>
</div>
<div>Global Signalling Settings:</div>
<div>---------------------------</div>
<div> Codecs: 0x4 (ulaw)</div>
<div> Codec Order: ulaw:20</div>
<div> Relax DTMF: No</div>
<div> RFC2833 Compensation: No</div>
<div> Symmetric RTP: No</div>
<div> Compact SIP headers: No</div>
<div> RTP Keepalive: 0 (Disabled)</div>
<div> RTP Timeout: 0 (Disabled)</div>
<div> RTP Hold Timeout: 0 (Disabled)</div>
<div> MWI NOTIFY mime type:
application/simple-message-summary</div>
<div>
DNS SRV lookup: Yes</div>
<div> Pedantic SIP support: Yes</div>
<div> Reg. min duration 60 secs</div>
<div> Reg. max duration: 3600 secs</div>
<div> Reg. default duration: 120 secs</div>
<div> Outbound reg. timeout: 20 secs</div>
<div> Outbound reg. attempts: 0</div>
<div> Notify ringing state: Yes</div>
<div> Include CID: No</div>
<div> Notify hold state: No</div>
<div> SIP Transfer mode: open</div>
<div> Max Call Bitrate: 384 kbps</div>
<div> Auto-Framing: No</div>
<div> Outb. proxy: <not set></div>
<div> Session Timers: Accept</div>
<div> Session Refresher: uas</div>
<div> Session Expires: 1800 secs</div>
<div>
Session Min-SE: 90 secs</div>
<div> Timer T1: 500</div>
<div> Timer T1 minimum: 100</div>
<div> Timer B: 32000</div>
<div> No premature media: Yes</div>
<div> Max forwards: 70</div>
<div><br>
</div>
<div>Default Settings:</div>
<div>-----------------</div>
<div> Allowed transports: UDP</div>
<div> Outbound transport: UDP</div>
<div> Context: entrada</div>
<div> Force rport: Yes</div>
<div> DTMF: rfc2833</div>
<div> Qualify: 0</div>
<div> Use ClientCode: No</div>
<div> Progress inband: Never</div>
<div> Language:</div>
<div> MOH Interpret: default</div>
<div> MOH Suggest:</div>
<div> Voice Mail Extension: asterisk</div>
<div><br>
</div>
</div>
<div><br>
</div>
<div>
<div>servidor*CLI> sip show peer 9960</div>
<div><br>
</div>
<div><br>
</div>
<div> * Name : 9960</div>
<div>
Secret : <Set></div>
<div> MD5Secret : <Not set></div>
<div> Remote Secret: <Not set></div>
<div> Context : ramais</div>
<div> Subscr.Cont. : <Not set></div>
<div> Language :</div>
<div> AMA flags : Unknown</div>
<div> Transfer mode: open</div>
<div> CallingPres : Presentation Allowed, Not Screened</div>
<div> Callgroup : 1</div>
<div> Pickupgroup : 1</div>
<div> MOH Suggest :</div>
<div> Mailbox : 9960</div>
<div> VM Extension : asterisk</div>
<div> LastMsgsSent : 32767/65535</div>
<div> Call limit : 0</div>
<div> Max forwards : 0</div>
<div> Dynamic : Yes</div>
<div> Callerid : "" <9960></div>
<div>
MaxCallBR : 384 kbps</div>
<div> Expire : 2311</div>
<div> Insecure : no</div>
<div> Force rport : Yes</div>
<div> ACL : No</div>
<div> DirectMedACL : No</div>
<div> T.38 support : No</div>
<div> T.38 EC mode : Unknown</div>
<div> T.38 MaxDtgrm: -1</div>
<div> DirectMedia : No</div>
<div> PromiscRedir : No</div>
<div> User=Phone : No</div>
<div> Video Support: No</div>
<div> Text Support : No</div>
<div> Ign SDP ver : No</div>
<div> Trust RPID : No</div>
<div> Send RPID : No</div>
<div> Subscriptions: Yes</div>
<div> Overlap dial : Yes</div>
<div> DTMFmode : rfc2833</div>
<div> Timer T1 : 500</div>
<div> Timer B : 32000</div>
<div> ToHost :</div>
<div>
Addr->IP : <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a></div>
<div> Defaddr->IP : (null)</div>
<div> Prim.Transp. : UDP</div>
<div> Allowed.Trsp : UDP</div>
<div> Def. Username: 9960</div>
<div> SIP Options : (none)</div>
<div> Codecs : 0xe (gsm|ulaw|alaw)</div>
<div> Codec Order : (ulaw:20,alaw:20,gsm:20)</div>
<div> Auto-Framing : No</div>
<div> Status : OK (201 ms)</div>
<div> Useragent : X-Lite release 1011s stamp 41150</div>
<div> Reg. Contact :
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a></div>
<div> Qualify Freq : 60000 ms</div>
<div> Sess-Timers : Accept</div>
<div> Sess-Refresh : uas</div>
<div> Sess-Expires : 1800 secs</div>
<div>
Min-Sess : 90 secs</div>
<div> RTP Engine : asterisk</div>
<div> Parkinglot :</div>
<div> Use Reason : No</div>
<div> Encryption : No</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>servidor*CLI> sip show peer 9961</div>
<div><br>
</div>
<div><br>
</div>
<div> * Name : 9961</div>
<div> Secret : <Set></div>
<div> MD5Secret : <Not set></div>
<div> Remote Secret: <Not set></div>
<div> Context : ramais</div>
<div> Subscr.Cont. : <Not set></div>
<div> Language :</div>
<div> AMA flags : Unknown</div>
<div> Transfer mode: open</div>
<div> CallingPres : Presentation Allowed, Not Screened</div>
<div> Callgroup : 1</div>
<div> Pickupgroup : 1</div>
<div> MOH Suggest :</div>
<div> Mailbox : 9961</div>
<div> VM Extension : asterisk</div>
<div> LastMsgsSent : 32767/65535</div>
<div> Call limit : 0</div>
<div> Max forwards : 0</div>
<div> Dynamic : Yes</div>
<div> Callerid : "" <9961></div>
<div> MaxCallBR : 384 kbps</div>
<div> Expire : 3065</div>
<div> Insecure : no</div>
<div> Force rport : Yes</div>
<div> ACL : No</div>
<div> DirectMedACL : No</div>
<div> T.38 support : No</div>
<div> T.38 EC mode : Unknown</div>
<div> T.38 MaxDtgrm: -1</div>
<div> DirectMedia : No</div>
<div> PromiscRedir : No</div>
<div> User=Phone : No</div>
<div> Video Support: No</div>
<div> Text Support : No</div>
<div> Ign SDP ver : No</div>
<div> Trust RPID : No</div>
<div> Send RPID : No</div>
<div> Subscriptions: Yes</div>
<div> Overlap dial : Yes</div>
<div> DTMFmode : rfc2833</div>
<div>
Timer T1 : 500</div>
<div> Timer B : 32000</div>
<div> ToHost :</div>
<div> Addr->IP : <a moz-do-not-send="true"
href="http://200.193.70.93:24477">200.193.70.93:24477</a></div>
<div> Defaddr->IP : (null)</div>
<div> Prim.Transp. : UDP</div>
<div> Allowed.Trsp : UDP</div>
<div> Def. Username: 9961</div>
<div> SIP Options : (none)</div>
<div> Codecs : 0xe (gsm|ulaw|alaw)</div>
<div> Codec Order : (ulaw:20,alaw:20,gsm:20)</div>
<div> Auto-Framing : No</div>
<div> Status : OK (179 ms)</div>
<div> Useragent : X-Lite release 1011s stamp 41150</div>
<div> Reg. Contact :
<a class="moz-txt-link-abbreviated" href="mailto:sip:9961@200.193.70.93:24477;rinstance=0f6639093db4a1d7">sip:9961@200.193.70.93:24477;rinstance=0f6639093db4a1d7</a></div>
<div> Qualify Freq : 60000 ms</div>
<div>
Sess-Timers : Accept</div>
<div> Sess-Refresh : uas</div>
<div> Sess-Expires : 1800 secs</div>
<div> Min-Sess : 90 secs</div>
<div> RTP Engine : asterisk</div>
<div> Parkinglot :</div>
<div> Use Reason : No</div>
<div> Encryption : No</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>servidor*CLI> sip set debug peer 9960</div>
<div>SIP Debugging Enabled for IP: 200.193.70.93</div>
<div>Reliably Transmitting (NAT) to <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>OPTIONS
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK6d96589d;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "asterisk" <<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@192.168.1.5">sip:asterisk@192.168.1.5</a>>;tag=as1db12108</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:asterisk@192.168.1.5:5060">sip:asterisk@192.168.1.5:5060</a>></div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060">49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060</a></div>
<div>CSeq: 102 OPTIONS</div>
<div>User-Agent: Asterisk PBX 1.8.11.0</div>
<div>Date: Mon, 28 Oct 2013 17:37:40 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><br>
</div>
<div>---</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK6d96589d;rport=5060;received=189.114.206.85</div>
<div>Contact: <sip:<a moz-do-not-send="true"
href="http://10.0.0.100:14069">10.0.0.100:14069</a>></div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=b1594d26</div>
<div>From: "asterisk"<<a moz-do-not-send="true"
href="mailto:sip%3Aasterisk@192.168.1.5">sip:asterisk@192.168.1.5</a>>;tag=as1db12108</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060">49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060</a></div>
<div>CSeq: 102 OPTIONS</div>
<div>Accept: application/sdp</div>
<div>Accept-Language: en</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO</div>
<div>User-Agent: X-Lite release 1011s stamp 41150</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><-------------></div>
<div>--- (12 headers 0 lines) ---</div>
<div>Really destroying SIP dialog '<a moz-do-not-send="true"
href="http://49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060">49863d0e131380840c553e5d7854a8a2@192.168.1.5:5060</a>'
Method: OPTIONS</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div><br>
</div>
<div><br>
</div>
<div><-------------></div>
<div> == Using SIP RTP CoS mark 5</div>
<div>
-- Executing [9960@ramais:1] Dial("SIP/9961-00000008",
"sip/9960") in new stack</div>
<div> == Using SIP RTP CoS mark 5</div>
<div>Audio is at 13418</div>
<div>Adding codec 0x4 (ulaw) to SDP</div>
<div>
Adding codec 0x8 (alaw) to SDP</div>
<div>Adding codec 0x2 (gsm) to SDP</div>
<div>Adding non-codec 0x1 (telephone-event) to SDP</div>
<div>Reliably Transmitting (NAT) to <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>INVITE
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "9961" <<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 1.8.11.0</div>
<div>Date: Mon, 28 Oct 2013 17:37:41 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH</div>
<div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 282</div>
<div><br>
</div>
<div>v=0</div>
<div>o=root 2034619578 2034619578 IN IP4 192.168.1.5</div>
<div>s=Asterisk PBX 1.8.11.0</div>
<div>c=IN IP4 192.168.1.5</div>
<div>t=0 0</div>
<div>m=audio 13418 RTP/AVP 0 8 3 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:3 GSM/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>
a=ptime:20</div>
<div>a=sendrecv</div>
<div><br>
</div>
<div>---</div>
<div> -- Called sip/9960</div>
<div>Retransmitting #1 (NAT) to <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>INVITE
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "9961" <<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 1.8.11.0</div>
<div>Date: Mon, 28 Oct 2013 17:37:41 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH</div>
<div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 282</div>
<div><br>
</div>
<div>v=0</div>
<div>o=root 2034619578 2034619578 IN IP4 192.168.1.5</div>
<div>s=Asterisk PBX 1.8.11.0</div>
<div>c=IN IP4 192.168.1.5</div>
<div>t=0 0</div>
<div>m=audio 13418 RTP/AVP 0 8 3 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:3 GSM/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>
a=ptime:20</div>
<div>a=sendrecv</div>
<div><br>
</div>
<div>---</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div>SIP/2.0 100 Trying</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>From: "9961" <<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><-------------></div>
<div>--- (7 headers 0 lines) ---</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div>SIP/2.0 180 Ringing</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85</div>
<div>Contact:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=4e3cab19</div>
<div>From: "9961"<<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: X-Lite release 1011s stamp 41150</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><-------------></div>
<div>--- (9 headers 0 lines) ---</div>
<div>list_route: hop:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div> -- SIP/9960-00000009 is ringing</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85</div>
<div>Contact:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=4e3cab19</div>
<div>From: "9961"<<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 INVITE</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO</div>
<div>Content-Type: application/sdp</div>
<div>User-Agent: X-Lite release 1011s stamp 41150</div>
<div>Content-Length: 180</div>
<div><br>
</div>
<div>v=0</div>
<div>o=- 4 2 IN IP4 10.0.0.100</div>
<div>s=CounterPath X-Lite 3.0</div>
<div>c=IN IP4 10.0.0.100</div>
<div>t=0 0</div>
<div>m=audio 6270 RTP/AVP 0 8 101</div>
<div>a=fmtp:101 0-15</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=sendrecv</div>
<div><-------------></div>
<div>--- (11 headers 9 lines) ---</div>
<div>Found RTP audio format 0</div>
<div>Found RTP audio format 8</div>
<div>Found RTP audio format 101</div>
<div>Found audio description format telephone-event for ID 101</div>
<div>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)</div>
<div>Non-codec capabilities (dtmf): us - 0x1
(telephone-event|), peer - 0x1 (telephone-event|), combined
- 0x1 (telephone-event|)</div>
<div>Peer audio RTP is at port <a moz-do-not-send="true"
href="http://10.0.0.100:6270">10.0.0.100:6270</a></div>
<div>list_route: hop:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>set_destination: Parsing
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>
for address/port to send to</div>
<div>
set_destination: set destination to <a
moz-do-not-send="true" href="http://200.193.70.93:2683">200.193.70.93:2683</a></div>
<div>Transmitting (NAT) to <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>ACK
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK3dcc2c0e;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "9961" <<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=4e3cab19</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 102 ACK</div>
<div>User-Agent: Asterisk PBX 1.8.11.0</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><br>
</div>
<div>---</div>
<div> -- SIP/9960-00000009 answered SIP/9961-00000008</div>
<div> -- Locally bridging SIP/9961-00000008 and
SIP/9960-00000009</div>
<div>[Oct 28 15:37:57] WARNING[2072]: chan_sip.c:3641
retrans_pkt: Retransmission timeout reached on transmission
ZmE1NmM4MDM5YjY1MzIxMDVkNTA4NWJkMTA0YTBlM2M. for seqno 2
(Critical Response) -- See <a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a></div>
<div>Packet timed out after 11455ms with no response</div>
<div>[Oct 28 15:37:57] WARNING[2072]: chan_sip.c:3670
retrans_pkt: Hanging up call
ZmE1NmM4MDM5YjY1MzIxMDVkNTA4NWJkMTA0YTBlM2M. - no reply to
our critical packet (see <a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).</div>
<div>Scheduling destruction of SIP dialog '<a
moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a>'
in 32320 ms (Method: INVITE)</div>
<div>set_destination: Parsing
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>
for address/port to send to</div>
<div>set_destination: set destination to <a
moz-do-not-send="true" href="http://200.193.70.93:2683">200.193.70.93:2683</a></div>
<div>Reliably Transmitting (NAT) to <a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>BYE
<a class="moz-txt-link-abbreviated" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898">sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK3ecf897d;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "9961" <<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=4e3cab19</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 103 BYE</div>
<div>User-Agent: Asterisk PBX 1.8.11.0</div>
<div>X-Asterisk-HangupCause: Normal Clearing</div>
<div>X-Asterisk-HangupCauseCode: 16</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><br>
</div>
<div>
---</div>
<div> == Spawn extension (ramais, 9960, 1) exited non-zero on
'SIP/9961-00000008'</div>
<div><br>
</div>
<div><--- SIP read from UDP:<a moz-do-not-send="true"
href="http://200.193.70.93:2683">200.193.70.93:2683</a>
---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP
192.168.1.5:5060;branch=z9hG4bK3ecf897d;rport=5060;received=189.114.206.85</div>
<div>Contact:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a></div>
<div>To:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898"><sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></a>;tag=4e3cab19</div>
<div>From: "9961"<<a moz-do-not-send="true"
href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a></div>
<div>CSeq: 103 BYE</div>
<div>User-Agent: X-Lite release 1011s stamp 41150</div>
<div>Content-Length: 0</div>
<div><br>
</div>
<div><-------------></div>
<div>--- (9 headers 0 lines) ---</div>
<div>Really destroying SIP dialog '<a moz-do-not-send="true"
href="http://362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060">362787273f6e6d56131c13e523f5ea68@192.168.1.5:5060</a>'
Method: INVITE</div>
<div>servidor*CLI></div>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">Em 28 de outubro de 2013 15:31,
Fernando - CIO - NextBilling IP Solutions <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:fernando@nextbilling.com.br" target="_blank">fernando@nextbilling.com.br</a>></span>
escreveu:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#333333"> no CLI do asterisk: <b>sip
show settings</b><br>
no CLI do asterisk: <b>sip show peer 9960</b><br>
no CLI do asterisk: <b>sip set debug peer 9960</b><br>
<br>
Faz a ligação e depois posta toda a saida aqui na lista.
fica mais facil te ajudar.<br>
<br>
<div>Em 28-10-2013 15:25, Fernando Trilha escreveu:<br>
</div>
<div>
<div class="h5">
<blockquote type="cite">
<div dir="ltr">Hudson, esta assim:
<div><br>
</div>
<div>
<div>[9960]</div>
<div>type=friend</div>
<div>secret=XXXXX</div>
<div>host=dynamic</div>
<div>mailbox=9960</div>
<div>context=ramais</div>
<div>callerid=9960</div>
<div>directmedia=no</div>
<div>dtmfmode=rfc2833</div>
<div>disallow=all</div>
<div>allow=ulaw</div>
<div>allow=alaw</div>
<div>allow=gsm</div>
<div>qualify=yes</div>
<div>callgroup=1</div>
<div>pickupgroup=1</div>
<div>canreinvite=no</div>
<div>nat=yes</div>
<div>externrefresh=10</div>
<div>externhost=189.114.206.85</div>
<div>localnet=<a moz-do-not-send="true"
href="http://192.168.1.0/255.255.255.0"
target="_blank">192.168.1.0/255.255.255.0</a></div>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote"> Em 28 de outubro de 2013
15:23, Hudson Cardoso <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:hudsoncardoso@hotmail.com"
target="_blank">hudsoncardoso@hotmail.com</a>></span>
escreveu:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">
<div>
<div dir="ltr"><font style="font-size:12pt"
face="Arial" size="3">Coloca</font>
<div><font style="font-size:12pt"
face="Arial" size="3">nat=yes no teu
sip conf.<br>
</font>
<div><br>
<br>
<pre style="line-height:17px;color:rgb(42,42,42);white-space:normal">Hudson
(048) 8413-7000
Para quem nao cre, nenhuma prova converte,Para aquele que cre, nenhuma prova precisa. </pre>
<br>
<br>
</div>
<div>
<hr>Date: Mon, 28 Oct 2013 12:38:55
-0200
<div>
<div><br>
From: <a moz-do-not-send="true"
href="mailto:ftrilha@gmail.com"
target="_blank">ftrilha@gmail.com</a><br>
To: <a moz-do-not-send="true"
href="mailto:asteriskbrasil@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>
Subject: Re: [AsteriskBrasil]
Ligação entre ramais muda<br>
<br>
<div dir="ltr">Agora deu este
erro:
<div> <br>
</div>
<div>
<div>== Using SIP RTP CoS mark
5</div>
<div> -- Called SIP/9961</div>
<div> -- SIP/9961-00000004
is ringing</div>
<div> -- SIP/9961-00000004
answered SIP/9960-00000003</div>
<div> -- Locally bridging
SIP/9960-00000003 and
SIP/9961-00000004</div>
<div> == Spawn extension
(ramais, 9961, 1) exited
non-zero on
'SIP/9960-00000003'</div>
<div>[Oct 28 12:35:30]
WARNING[1629]:
chan_sip.c:3641 retrans_pkt:
Retransmission timeout
reached on transmission
ZWRlMDgyYmUzMWUyNzg1M2I5NzJjNWM4ZWJhOTRhNTk.
for seqno 2 (Critical
Response) -- See <a
moz-do-not-send="true"
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
12928ms with no response</div>
</div>
<div><br>
</div>
</div>
<div><br>
<br>
<div>Em 28 de outubro de 2013
12:05, Hudson Cardoso <span
dir="ltr"><<a
moz-do-not-send="true"
href="mailto:hudsoncardoso@hotmail.com"
target="_blank">hudsoncardoso@hotmail.com</a>></span>
escreveu:<br>
<blockquote
style="border-left:1px #ccc
solid;padding-left:1ex">
<div>
<div dir="ltr"><font
style="font-size:12pt"
face="Arial" size="3">
Testa com iax, se
funcionar, é rtp com
problemas.<br>
</font><br>
<br>
<pre style="line-height:17px;color:rgb(42,42,42);white-space:normal">Hudson
(048) 8413-7000
Para quem nao cre, nenhuma prova converte,Para aquele que cre, nenhuma prova precisa. </pre>
<br>
<br>
<div>
<hr>Date: Mon, 28 Oct
2013 12:03:13 -0200<br>
From: <a
moz-do-not-send="true"
href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br>
To: <a
moz-do-not-send="true"
href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>
Subject: Re:
[AsteriskBrasil]
Ligação entre ramais
muda
<div>
<div><br>
<br>
<div dir="ltr">
Marcelo, fiz a
alteração mas
continua a mesma
coisa.</div>
<div><br>
<br>
<div>Em 26 de
outubro de
2013 09:57,
Marcelo Terres
<span
dir="ltr"><<a
moz-do-not-send="true" href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a>></span>
escreveu:<br>
<blockquote
style="border-left:1px
#ccc
solid;padding-left:1ex">Seta
o
directmedia=no
para os dois
ramais no
sip.conf,
registra eles<br>
novamente e
testa para ver
se muda algo.<br>
<div><br>
[]s<br>
Marcelo H.
Terres<br>
<a
moz-do-not-send="true"
href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a><br>
<a
moz-do-not-send="true"
href="http://mundoopensource.blogspot.com" target="_blank">http://mundoopensource.blogspot.com</a><br>
<a
moz-do-not-send="true"
href="http://biertasters.blogspot.com" target="_blank">http://biertasters.blogspot.com</a><br>
<a
moz-do-not-send="true"
href="http://twitter.com/mhterres" target="_blank">http://twitter.com/mhterres</a><br>
<br>
<br>
</div>
Em 26 de
outubro de
2013 09:35,
Fernando
Trilha <<a
moz-do-not-send="true" href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a>>
escreveu:<br>
<div>
<div>> Sim,
os mesmo, gsm,
allaw e ullaw.<br>
><br>
><br>
> Em 26 de
outubro de
2013 09:28,
Marcelo Terres
<<a
moz-do-not-send="true"
href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a>><br>
> escreveu:<br>
><br>
>> E os
codecs, são os
mesmos ?<br>
>><br>
>> Quais
codecs tu
configurou no
sip.conf e no
zoiper?<br>
>><br>
>> []s<br>
>>
Marcelo H.
Terres<br>
>> <a
moz-do-not-send="true"
href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a><br>
>> <a
moz-do-not-send="true"
href="http://mundoopensource.blogspot.com" target="_blank">http://mundoopensource.blogspot.com</a><br>
>> <a
moz-do-not-send="true"
href="http://biertasters.blogspot.com" target="_blank">http://biertasters.blogspot.com</a><br>
>> <a
moz-do-not-send="true"
href="http://twitter.com/mhterres" target="_blank">http://twitter.com/mhterres</a><br>
>><br>
>><br>
>> Em 25
de outubro de
2013 20:18,
Fernando
Trilha <<a
moz-do-not-send="true" href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a>><br>
>>
escreveu:<br>
>> >
Pessoal tenho
um asterisk
instalado,
apenas para
ligações entre
ramais,<br>
>> >
toca, atende
mas mudo os
dois lados.<br>
>> >
Uso o zoiper
configurado em
dois
smartphones
com protocolo
SIP, nao da<br>
>> >
erro<br>
>> >
no CLI.<br>
>> ><br>
>> >
--<br>
>> >
Atte.<br>
>> >
Fernando
Trilha<br>
>> ><br>
>> ><br>
>> >
_______________________________________________<br>
>> >
KHOMP:
completa linha
de placas
externas FXO,
FXS, GSM e E1;<br>
>> >
Media Gateways
de 1 a 64 E1s
para SIP com
R2, ISDN e
SS7;<br>
>> >
Intercomunicadores
para acesso
remoto via
rede IP.
Conheça em<br>
>> >
<a
moz-do-not-send="true"
href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
>> >
_______________________________________________<br>
>> >
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>
>> >
Channel Bank –
Appliance
Asterisk -
Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.<br>
>> >
_______________________________________________<br>
>> >
Para remover
seu email
desta lista,
basta enviar
um email em
branco para<br>
>> >
<a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
>><br>
>>
_______________________________________________<br>
>>
KHOMP:
completa linha
de placas
externas FXO,
FXS, GSM e E1;<br>
>> Media
Gateways de 1
a 64 E1s para
SIP com R2,
ISDN e SS7;<br>
>>
Intercomunicadores
para acesso
remoto via
rede IP.
Conheça em<br>
>> <a
moz-do-not-send="true"
href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
>>
_______________________________________________<br>
>>
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>
>>
Channel Bank –
Appliance
Asterisk -
Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.<br>
>>
_______________________________________________<br>
>> Para
remover seu
email desta
lista, basta
enviar um
email em
branco para<br>
>> <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
><br>
><br>
><br>
><br>
> --<br>
> Atte.<br>
> Fernando
Trilha<br>
> Analista
de Suporte<br>
> 8414 -
6008<br>
> <a
moz-do-not-send="true"
href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br>
>
::Soluções em
informatica e
redes
corporativas::<br>
><br>
>
_______________________________________________<br>
> KHOMP:
completa linha
de placas
externas FXO,
FXS, GSM e E1;<br>
> 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
moz-do-not-send="true" href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
>
_______________________________________________<br>
> 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>
> Channel
Bank –
Appliance
Asterisk -
Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.<br>
>
_______________________________________________<br>
> Para
remover seu
email desta
lista, basta
enviar um
email em
branco para<br>
> <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
</div>
</div>
<br>
_______________________________________________<br>
KHOMP:
completa linha
de placas
externas FXO,
FXS, GSM e E1;<br>
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
moz-do-not-send="true" href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
_______________________________________________<br>
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>
Channel Bank –
Appliance
Asterisk -
Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.<br>
_______________________________________________<br>
Para remover
seu email
desta lista,
basta enviar
um email em
branco para <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr">Atte.<br>
Fernando
Trilha<br>
Analista de
Suporte
<div>8414 -
6008<br>
<a
moz-do-not-send="true"
href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br>
</div>
<div>::Soluções
em informatica
e redes
corporativas::</div>
</div>
</div>
<br>
</div>
</div>
_______________________________________________
KHOMP:
completa linha de
placas externas FXO,
FXS, GSM e E1; Media
Gateways de 1 a 64 E1s
para SIP com R2, ISDN
e SS7;
Intercomunicadores
para acesso remoto via
rede IP. Conhe�a em <a
moz-do-not-send="true" href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.
_______________________________________________
ALIGERA
� Fabricante nacional
de Gateways SIP-E1
para R2, ISDN e SS7.
Placas de 1E1, 2E1,
4E1 e 8E1 para PCI ou
PCI Express. Channel
Bank � Appliance
Asterisk - Acesse <a
moz-do-not-send="true" href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.
_______________________________________________
Para
remover seu email
desta lista, basta
enviar um email em
branco para <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div>
</div>
</div>
<br>
_______________________________________________<br>
KHOMP: completa linha de
placas externas FXO, FXS,
GSM e E1;<br>
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
moz-do-not-send="true"
href="http://www.Khomp.com"
target="_blank">www.Khomp.com</a>.<br>
_______________________________________________<br>
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>
Channel Bank – Appliance
Asterisk - Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br"
target="_blank">www.aligera.com.br</a>.<br>
_______________________________________________<br>
Para remover seu email desta
lista, basta enviar um email
em branco para <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr">Atte.<br>
Fernando Trilha<br>
Analista de Suporte
<div>8414 - 6008<br>
<a moz-do-not-send="true"
href="mailto:ftrilha@gmail.com"
target="_blank">ftrilha@gmail.com</a><br>
</div>
<div>::Soluções em informatica
e redes corporativas::</div>
</div>
</div>
<br>
_______________________________________________
KHOMP: completa linha de placas
externas FXO, FXS, GSM e E1; Media
Gateways de 1 a 64 E1s para SIP
com R2, ISDN e SS7;
Intercomunicadores para acesso
remoto via rede IP. Conhe�a em <a
moz-do-not-send="true"
href="http://www.Khomp.com"
target="_blank">www.Khomp.com</a>.
_______________________________________________
ALIGERA � Fabricante nacional de
Gateways SIP-E1 para R2, ISDN e
SS7. Placas de 1E1, 2E1, 4E1 e 8E1
para PCI ou PCI Express. Channel
Bank � Appliance Asterisk - Acesse
<a moz-do-not-send="true"
href="http://www.aligera.com.br"
target="_blank">www.aligera.com.br</a>.
_______________________________________________
Para remover seu email desta
lista, basta enviar um email em
branco para <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div>
</div>
</div>
</div>
</div>
</div>
<br>
_______________________________________________<br>
KHOMP: completa linha de placas externas FXO,
FXS, GSM e E1;<br>
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 moz-do-not-send="true"
href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.<br>
_______________________________________________<br>
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>
Channel Bank – Appliance Asterisk - Acesse <a
moz-do-not-send="true"
href="http://www.aligera.com.br"
target="_blank">www.aligera.com.br</a>.<br>
_______________________________________________<br>
Para remover seu email desta lista, basta
enviar um email em branco para <a
moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr">Atte.<br>
Fernando Trilha<br>
Analista de Suporte
<div>8414 - 6008<br>
<a moz-do-not-send="true"
href="mailto:ftrilha@gmail.com"
target="_blank">ftrilha@gmail.com</a><br>
</div>
<div>::Soluções em informatica e redes
corporativas::</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
Intercomunicadores para acesso remoto via rede IP. Conheça em <a moz-do-not-send="true" href="http://www.Khomp.com" target="_blank">www.Khomp.com</a>.
_______________________________________________
ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
Channel Bank – Appliance Asterisk - Acesse <a moz-do-not-send="true" href="http://www.aligera.com.br" target="_blank">www.aligera.com.br</a>.
_______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para <a moz-do-not-send="true" href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org" target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></pre>
</blockquote>
<br>
</div>
</div>
</div>
<br>
_______________________________________________<br>
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;<br>
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 moz-do-not-send="true" href="http://www.Khomp.com"
target="_blank">www.Khomp.com</a>.<br>
_______________________________________________<br>
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>
Channel Bank – Appliance Asterisk - Acesse <a
moz-do-not-send="true" href="http://www.aligera.com.br"
target="_blank">www.aligera.com.br</a>.<br>
_______________________________________________<br>
Para remover seu email desta lista, basta enviar um email em
branco para <a moz-do-not-send="true"
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr">Atte.<br>
Fernando Trilha<br>
Analista de Suporte
<div>8414 - 6008<br>
<a moz-do-not-send="true" href="mailto:ftrilha@gmail.com"
target="_blank">ftrilha@gmail.com</a><br>
</div>
<div>::Soluções em informatica e redes corporativas::</div>
</div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
Intercomunicadores para acesso remoto via rede IP. Conheça em <a class="moz-txt-link-abbreviated" href="http://www.Khomp.com">www.Khomp.com</a>.
_______________________________________________
ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
Channel Bank – Appliance Asterisk - Acesse <a class="moz-txt-link-abbreviated" href="http://www.aligera.com.br">www.aligera.com.br</a>.
_______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para <a class="moz-txt-link-abbreviated" href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></pre>
</blockquote>
<br>
</body>
</html>