<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 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 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 : sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898</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 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 : sip:9961@200.193.70.93:24477;rinstance=0f6639093db4a1d7</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 href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>OPTIONS sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 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 href="mailto:sip%3Aasterisk@192.168.1.5">sip:asterisk@192.168.1.5</a>>;tag=as1db12108</div>
<div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>Contact: <<a href="http://sip:asterisk@192.168.1.5:5060">sip:asterisk@192.168.1.5:5060</a>></div><div>Call-ID: <a 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 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 href="http://10.0.0.100:14069">10.0.0.100:14069</a>></div><div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=b1594d26</div><div>From: "asterisk"<<a href="mailto:sip%3Aasterisk@192.168.1.5">sip:asterisk@192.168.1.5</a>>;tag=as1db12108</div>
<div>Call-ID: <a 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 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 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 href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div>
<div>INVITE sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 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 href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>Contact: <<a href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div><div>Call-ID: <a 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 href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div><div>INVITE sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 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 href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>Contact: <<a href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div><div>Call-ID: <a 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 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: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>From: "9961" <<a href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div><div>Call-ID: <a 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 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: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19</div>
<div>From: "9961"<<a href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div><div>Call-ID: <a 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: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div>
<div> -- SIP/9960-00000009 is ringing</div><div><br></div><div><--- SIP read from UDP:<a 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: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19</div><div>From: "9961"<<a href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>Call-ID: <a 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 href="http://10.0.0.100:6270">10.0.0.100:6270</a></div>
<div>list_route: hop: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>set_destination: Parsing <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898> for address/port to send to</div><div>
set_destination: set destination to <a href="http://200.193.70.93:2683">200.193.70.93:2683</a></div><div>Transmitting (NAT) to <a href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div><div>ACK sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 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 href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19</div><div>Contact: <<a href="http://sip:9961@192.168.1.5:5060">sip:9961@192.168.1.5:5060</a>></div><div>Call-ID: <a 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 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 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 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 <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898> for address/port to send to</div>
<div>set_destination: set destination to <a href="http://200.193.70.93:2683">200.193.70.93:2683</a></div><div>Reliably Transmitting (NAT) to <a href="http://200.193.70.93:2683">200.193.70.93:2683</a>:</div><div>BYE sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 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 href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div>
<div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19</div><div>Call-ID: <a 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 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: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898></div><div>To: <sip:9960@200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19</div>
<div>From: "9961"<<a href="mailto:sip%3A9961@192.168.1.5">sip:9961@192.168.1.5</a>>;tag=as1be09e45</div><div>Call-ID: <a 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 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 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 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 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 href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br>
To: <a 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 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 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 href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br>
To: <a 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 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 href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a><br>
<a href="http://mundoopensource.blogspot.com" target="_blank">http://mundoopensource.blogspot.com</a><br>
<a href="http://biertasters.blogspot.com" target="_blank">http://biertasters.blogspot.com</a><br>
<a 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 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 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 href="mailto:mhterres@gmail.com" target="_blank">mhterres@gmail.com</a><br>
>> <a href="http://mundoopensource.blogspot.com" target="_blank">http://mundoopensource.blogspot.com</a><br>
>> <a href="http://biertasters.blogspot.com" target="_blank">http://biertasters.blogspot.com</a><br>
>> <a 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 href="mailto:ftrilha@gmail.com" target="_blank">ftrilha@gmail.com</a><br></div><div>::Soluções em informatica e redes corporativas::</div>
</div>
</div>