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Ichi, como uso Asterisk Puro, não vou saber te ajudar. Mas essa
solução que encontrei foi essa e resolveu meu problema. Terás que
"futucar" no DialPlan do Asterisk.<br>
Abs..<br>
Em 31/10/2012 14:41, Ivan Maldonado Orosco escreveu:
<blockquote cite="mid:COL105-DS21C061B869D505BF5A3F7FD2610@phx.gbl"
type="cite">
<div dir="ltr">
<div style="FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE:
12pt">
<div>É meio extenso, segue:</div>
<div> </div>
<div>[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Audio is at
189.2.20.134 port 16664<br>
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec
0x4 (ulaw) to SDP<br>
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec
0x8 (alaw) to SDP<br>
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding
non-codec 0x1 (telephone-event) to SDP<br>
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Reliably
Transmitting (no NAT) to 189.47.46.151:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:97095313@189.47.46.151:5060">sip:97095313@189.47.46.151:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP
189.2.20.134:5060;branch=z9hG4bK4c8b3981;rport<br>
Max-Forwards: 70<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a><br>
Call-ID: <a moz-do-not-send="true"
href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq: 102 INVITE<br>
User-Agent: FPBX-2.8.1(1.6.2.17)<br>
Date: Tue, 30 Oct 2012 09:41:20 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 261</div>
<div> </div>
<div>v=0<br>
o=root 1238957108 1238957108 IN IP4 189.2.20.134<br>
s=Asterisk PBX 1.6.2.17<br>
c=IN IP4 189.2.20.134<br>
t=0 0<br>
m=audio 16664 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv</div>
<div> </div>
<div>---<br>
[Oct 30 07:41:20] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 100 Trying<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a><br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Content-Type:application/sdp<br>
Content-Length:0</div>
<div> </div>
<div><br>
<-------------><br>
[Oct 30 07:41:20] VERBOSE[8895] chan_sip.c: --- (8 headers 0
lines) ---<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 183 Session in progress<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849593660 1849593660 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: --- (10 headers
9 lines) ---<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio
format 0<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio
format 101<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio
description format PCMU for ID 0<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio
description format telephone-event for ID 101<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Capabilities: us
- 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Non-codec
capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)<br>
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Peer audio RTP
is at port 189.47.46.151:10000<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers
9 lines) ---<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio
format 0<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio
format 101<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio
description format PCMU for ID 0<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio
description format telephone-event for ID 101<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Capabilities: us
- 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Non-codec
capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Peer audio RTP
is at port 189.47.46.151:10000<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: list_route: hop:
<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination:
Parsing <a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a> for address/port
to send to<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination:
set destination to 189.47.46.151, port 5060<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Transmitting (no
NAT) to 189.47.46.151:5060:<br>
ACK <a class="moz-txt-link-freetext" href="sip:6720@189.47.46.151:5060">sip:6720@189.47.46.151:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP
189.2.20.134:5060;branch=z9hG4bK1c6b8c10;rport<br>
Max-Forwards: 70<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a><br>
Call-ID: <a moz-do-not-send="true"
href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq: 102 ACK<br>
User-Agent: FPBX-2.8.1(1.6.2.17)<br>
Content-Length: 0</div>
<div> </div>
<div><br>
---<br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers
9 lines) ---<br>
[Oct 30 07:41:34] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:34] VERBOSE[8895] chan_sip.c: --- (11 headers
9 lines) ---<br>
[Oct 30 07:41:36] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:36] VERBOSE[8895] chan_sip.c: --- (11 headers
9 lines) ---<br>
[Oct 30 07:41:40] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><br>
<-------------><br>
[Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: --- (10 headers
0 lines) ---<br>
[Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: Really
destroying SIP dialog <a moz-do-not-send="true"
href="mailto:%27644015366b7e382f266705505a200615@189.2.20.134%27">'644015366b7e382f266705505a200615@189.2.20.134'</a>
Method: OPTIONS<br>
[Oct 30 07:41:48] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
Via:SIP/2.0/UDP
189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
To:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:102 INVITE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Type:application/sdp<br>
Content-Length:209</div>
<div> </div>
<div>v=0<br>
o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
s=Session SDP<br>
c=IN IP4 189.47.46.151<br>
t=0 0<br>
m=audio 10000 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-16</div>
<div> </div>
<div><-------------><br>
[Oct 30 07:41:48] VERBOSE[8895] chan_sip.c: --- (11 headers
9 lines) ---<br>
[Oct 30 07:41:53] WARNING[18042] func_db.c: DB_DELETE
requires an argument, DB_DELETE(<family>/<key>)<br>
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: <br>
<--- SIP read from UDP:189.47.46.151:5060 ---><br>
BYE <a class="moz-txt-link-freetext" href="sip:100@189.2.20.134">sip:100@189.2.20.134</a> SIP/2.0<br>
Via:SIP/2.0/UDP
189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c<br>
From:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
To: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
<a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq:8 BYE<br>
Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060"><sip:6720@189.47.46.151:5060></a><br>
Max-Forwards:70<br>
User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
Content-Length:0</div>
<div> </div>
<div><br>
<-------------><br>
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: --- (10 headers
0 lines) ---<br>
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: Sending to
189.47.46.151 : 5060 (no NAT)<br>
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: <br>
<--- Transmitting (no NAT) to 189.47.46.151:5060 ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP
189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c;received=189.47.46.151<br>
From:
<a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060"><sip:97095313@189.47.46.151:5060></a>;tag=8ed4511d-742398<br>
To: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134"><sip:100@189.2.20.134></a>;tag=as57399d63<br>
Call-ID: <a moz-do-not-send="true"
href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
CSeq: 8 BYE<br>
Server: FPBX-2.8.1(1.6.2.17)<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Length: 0</div>
<div> </div>
<div><br>
<------------><br>
[Oct 30 07:41:54] VERBOSE[8895] chan_sip.c: Really
destroying SIP dialog <a moz-do-not-send="true"
href="mailto:%273abbce8a451fb58b6726e36623e98a89@189.2.20.134%27">'3abbce8a451fb58b6726e36623e98a89@189.2.20.134'</a>
Method: BYE<br>
[Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really
destroying SIP dialog <a moz-do-not-send="true"
href="mailto:%27CE40-3FA8-4674241856E1641BE1B1-193@SipHost%27">'CE40-3FA8-4674241856E1641BE1B1-193@SipHost'</a>
Method: REGISTER<br>
[Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really
destroying SIP dialog <a moz-do-not-send="true"
href="mailto:%27CE40-3FA8-4674242047F2EB3729E8-194@SipHost%27">'CE40-3FA8-4674242047F2EB3729E8-194@SipHost'</a>
Method: REGISTER<br>
</div>
<div>[]’s</div>
<div style="FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY:
'Calibri'; COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT:
normal; TEXT-DECORATION: none">
<div style="FONT: 10pt tahoma">
<div> </div>
<div style="BACKGROUND: #f5f5f5">
<div style="font-color: black"><b>From:</b> <a
moz-do-not-send="true" title="ivan.paes@gmail.com"
href="mailto:ivan.paes@gmail.com">Ivan Paes José</a>
</div>
<div><b>Sent:</b> Tuesday, October 30, 2012 12:20 AM</div>
<div><b>To:</b> <a moz-do-not-send="true"
title="asteriskbrasil@listas.asteriskbrasil.org"
href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</a>
</div>
<div><b>Subject:</b> Re: [AsteriskBrasil] DVG 6004S
St_VoipAnswering Timeout</div>
</div>
</div>
<div> </div>
</div>
<div style="FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY:
'Calibri'; COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT:
normal; TEXT-DECORATION: none">
<div>Olá!</div>
<div> </div>
<div>Consegues o debug do sip do CLI do asterisk?</div>
<br clear="all">
<font size="1"><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial">Atenciosamente,<br>
<br>
Ivan P</span></span><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial"></span></span><span
style="LINE-HEIGHT: normal; TEXT-TRANSFORM: none;
FONT-VARIANT: normal; FONT-STYLE: normal; TEXT-INDENT:
0px; LETTER-SPACING: normal; BORDER-COLLAPSE: separate;
FONT-FAMILY: 'Times New Roman'; WHITE-SPACE: normal;
COLOR: rgb(0,0,0); FONT-WEIGHT: normal; WORD-SPACING:
0px"><span style="FONT-FAMILY: arial">aes José<br>
</span></span><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial"></span></span><br>
<span style="LINE-HEIGHT: normal; TEXT-TRANSFORM: none;
FONT-VARIANT: normal; FONT-STYLE: normal; TEXT-INDENT:
0px; LETTER-SPACING: normal; BORDER-COLLAPSE: separate;
FONT-FAMILY: 'Times New Roman'; WHITE-SPACE: normal;
COLOR: rgb(0,0,0); FONT-WEIGHT: normal; WORD-SPACING:
0px"><span style="FONT-FAMILY: arial">Acadêmico de
Biblioteconomia - UFSC<br>
</span></span><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial">Técnico em Telecomunicações
- IFSC<br>
<br>
E-mail/MSN/GTalk:<span> </span><a
moz-do-not-send="true"
href="mailto:ivan.paes@gmail.com" target="_blank">ivan.paes@gmail.com</a><br>
Oi: +55 48 84291055<br>
</span></span><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial">Skype: ivanpaesjose<br>
</span></span><span style="LINE-HEIGHT: normal;
TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
style="FONT-FAMILY: arial">Palhoça - Santa Catarina -
Brasil<br>
<br>
*** Muito Importante *** NETiqueta<br>
Se repassar esta mensagem, por gentileza:<br>
* Apague todos os endereços que aparecem nele.<br>
* E, por opção de segurança endereçá-lo no Cco ou Bcc.<br>
Desta forma você estará protegendo a mim, seus amigos
e a você mesmo.<br>
Eu, juntamente com a campanha contra a propagação de
vírus agradecemos sinceramente.</span></span></font><br>
<br>
<br>
<div class="gmail_quote">Em 29 de outubro de 2012 17:28,
Ivan Maldonado Orosco <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:ivanorosco@hotmail.com" target="_blank">ivanorosco@hotmail.com</a>></span>
escreveu:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN:
0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class="gmail_quote">
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY: 'Calibri'; FONT-SIZE: 12pt">
<div>Boa tarde,</div>
<div> </div>
<div>Adquirimos um ata Dlink DVG 6004S e
configuramos o mesmo para utilizar nossas linhas
analógicas, no entanto, deparamos com um
problema aparentemente simples para se resolver,
no entanto, não conseguimos encontrar a solução.
O que acontece é que o ata recebe ligações e
transfere para os ramais SIP corretamente
(entrada 100% funcional), mas na execução de
discagem, o asterisk comanda a discagem, o ata
faz a discagem, o destino atende a ligação e os
dois pontos se falam, mas após 20 segundos da
discagem a ligação é derrubada pelo ata.</div>
<div> </div>
<div>Depurando as ações dos comandos do ata pelo
SLmon (programinha da Dlink), vimos que o mesmo
efetua todo o processo de discagem e envia por
último o comando “==13: VoipAnswering” (com a
ligação já atendida) e após 20 segundos ele
responde novamente com St_VoipAnswering Timeout
e derruba a ligação.</div>
<div> </div>
<div>Vejam o log gerado:</div>
<div> </div>
<div>16:58:00 [010667] 5: 6720=OFFERING<br>
16:58:00 [010667] 5: Get CallerId=100<br>
16:58:00 [010668] 5: Check Trunk FixLine<br>
16:58:00 [010668] 5: Hunting Trunk Line<br>
16:58:00 [010668] 0: Peer PTime=20 #2<br>
16:58:00 [010668] 0: Peer=<a
moz-do-not-send="true"
href="http://189.2.20.138:19292"
target="_blank">189.2.20.138:19292</a>, PT=0,
RecvOnly=0<br>
16:58:00 [010668] 0: TrunkPrefix=,
Dest=39064886, Dialno=39064886<br>
16:58:00 [010668] 0: FxoHookOff<br>
16:58:00 [010668] 0: SetInputGain(-2)<br>
16:58:00 [010668] 0: ==18:TrunkDialOut<br>
16:58:01 [010679] 0: DialOut(39064886)=0<br>
16:58:02 [010692] 0: RtpApiTalk[1,1],Peer=<a
moz-do-not-send="true"
href="http://189.2.20.138:19292"
target="_blank">189.2.20.138:19292</a>,PT=0,FX=2,NewOOB=1<br>
16:58:02 [010692] 0: Substatus=3<br>
16:58:11 [010782] 0: Fxo Still No RingTone, Talk<br>
16:58:11 [010782] 0: Fxo DialOut OK<br>
16:58:11 [010782] 5: 6720=ACCEPT<br>
16:58:11 [010783] 0: ==13:VoipAnswering</div>
<div><br>
16:58:33 [010996] 0: St_VoipAnswering Timeout<br>
16:58:33 [010996] 0: DSP_ch0_check=0<br>
16:58:33 [010996] 0: FxoHookOn<br>
16:58:33 [010996] 0: ==15:PlayBusyTone<br>
16:58:33 [010996] 0: DSP_ch0_check=0<br>
16:58:33 [010996] 0: FxoHookOn<br>
16:58:33 [010996] 0: ==3:Idle<br>
16:58:33 [010996] 0: SetInputGain(4)<br>
16:58:33 [010996] 0: SetFax(1)=0<br>
16:58:33 [010996] 0: 6720=DISCONNECT</div>
<div> </div>
<div>Já fiz todas as configurações possíveis no
ATA e não houve nenhum resultado diferente que
faça ele dar uma outra resposta após o
VoipAnswering que não seja Timeout.</div>
<div> </div>
<div>As configurações de meu tronco SIP são:</div>
<div>host=dynamic<br>
username=6720<br>
secret=<br>
type=friend<br>
qualify=yes<br>
canreinvite=yes<br>
dtmfmode=rfc2833<br>
alow=all<br>
</div>
<div>O ATA está ligado diretamente na internet sem
firewal na frente, assim como nosso asterisk
(para evitar qualquer problema relacionado a
liberação de portas)</div>
<div> </div>
<div>Alguém tem alguma idéia ?</div>
<div> </div>
<div>[]’s</div>
<div> </div>
<div><br>
</div>
<div> </div>
</div>
</div>
</div>
<br>
_______________________________________________<br>
KHOMP Inovação: External Board Series<br>
Módulos de 1/2 rack e 1U para todas as interfaces e
soluções Asterisk e FreeSWITCH.<br>
Tenha a External Series Experience na sua aplicação.
Visite <a moz-do-not-send="true"
href="http://www.khomp.com" target="_blank">www.khomp.com</a><br>
_______________________________________________<br>
DIGIVOICE Fabricante de Placas de Voz e Channel Bank<br>
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM<br>
Centro Treinamento - Curso de PABX IP - Asterisk -
Site <a moz-do-not-send="true"
href="http://www.digivoice.com.br" target="_blank">www.digivoice.com.br</a><br>
________<br>
YEALINK: Telefones IP e VídeoPhones IP com o melhor
custo/benefício do mercado.<br>
email: <a moz-do-not-send="true"
href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a>
| <a moz-do-not-send="true"
href="http://www.commlogik.com.br" target="_blank">www.commlogik.com.br</a>
| <a moz-do-not-send="true"
href="tel:%2811%29%205503-1011" value="+551155031011">(11)
5503-1011</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>
<p>
</p>
<hr>
_______________________________________________<br>
KHOMP Inovação: External Board Series<br>
Módulos de 1/2 rack e 1U para todas as interfaces e soluções
Asterisk e FreeSWITCH.<br>
Tenha a External Series Experience na sua aplicação. Visite
<a class="moz-txt-link-abbreviated" href="http://www.khomp.com">www.khomp.com</a><br>
_______________________________________________<br>
DIGIVOICE Fabricante de Placas de Voz e Channel Bank<br>
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM<br>
Centro Treinamento - Curso de PABX IP - Asterisk - Site
<a class="moz-txt-link-abbreviated" href="http://www.digivoice.com.br">www.digivoice.com.br</a><br>
________<br>
YEALINK: Telefones IP e VídeoPhones IP com o melhor
custo/benefício do mercado.<br>
email: <a class="moz-txt-link-abbreviated" href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a> | <a class="moz-txt-link-abbreviated" href="http://www.commlogik.com.br">www.commlogik.com.br</a> |
(11) 5503-1011<br>
______________________________________________<br>
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></div>
</div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
KHOMP Inovação: External Board Series
Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e FreeSWITCH.
Tenha a External Series Experience na sua aplicação. Visite <a class="moz-txt-link-abbreviated" href="http://www.khomp.com">www.khomp.com</a>
_______________________________________________
DIGIVOICE Fabricante de Placas de Voz e Channel Bank
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
Centro Treinamento - Curso de PABX IP - Asterisk - Site <a class="moz-txt-link-abbreviated" href="http://www.digivoice.com.br">www.digivoice.com.br</a>
________
YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do mercado.
email: <a class="moz-txt-link-abbreviated" href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a> | <a class="moz-txt-link-abbreviated" href="http://www.commlogik.com.br">www.commlogik.com.br</a> | (11) 5503-1011
______________________________________________
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>
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