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<DIV dir=ltr>
<DIV style="FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE: 12pt">
<DIV>Prezados, estou aqui ainda sofrendo pra fazer meu dlink funcionar, olhando
o verbose do cli observei o seguinte comportamento do asterisK</DIV>
<DIV> </DIV>
<DIV> == Using SIP RTP TOS bits 184<BR> == Using SIP RTP CoS mark
5<BR> == Begin MixMonitor Recording SIP/100-00002a07<BR>
-- Called 6720/97095313<BR> -- SIP/6720-00002a08 is making
progress passing it to SIP/100-00002a07<BR> --
SIP/6720-00002a08 answered SIP/100-00002a07<BR></DIV>
<DIV>ele já da o Answered mesmo sem chamar ! </DIV>
<DIV> </DIV>
<DIV>Não sei se tem algo a ver, mas olhando pelo sip debug vejo que no momento
que a linha está chamando, o dlink envia pacotes de INVITE apenas e o asterisk
não responde nada, será que é porque ele já deu Answered ?</DIV>
<DIV> </DIV>
<DIV>[]’s</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 style="FONT: 10pt tahoma">
<DIV> </DIV>
<DIV style="BACKGROUND: #f5f5f5">
<DIV style="font-color: black"><B>From:</B> <A title=asterisk@guilherme.eti.br
href="mailto:asterisk@guilherme.eti.br">Guilherme Rezende</A> </DIV>
<DIV><B>Sent:</B> Sunday, November 04, 2012 4:30 PM</DIV>
<DIV><B>To:</B> <A 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">Não
cheguei monitorar. Apenas precisa de uma solução rápida e fiz da forma que lhe
mostrei e está funcionando até hoje.<BR><BR>Em 01/11/2012 09:16, Ivan Maldonado
Orosco escreveu:
<BLOCKQUOTE cite=mid:COL105-DS787C8B473F9C938E4905AD2600@phx.gbl type="cite">
<DIV dir=ltr>
<DIV style="FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE: 12pt">
<DIV>Bom, ao menos é um caminho para eu pesquisar, mas Guilherme, o seu
problema era exatamente o mesmo ? você chegou a monitorar o seu dvg 6004/6008
e viu esse bemdito VoipAnswering e St_VoipAnswering Timeout ?</DIV>
<DIV> </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 title=asterisk@guilherme.eti.br
href="mailto:asterisk@guilherme.eti.br" moz-do-not-send="true">Guilherme
Rezende</A> </DIV>
<DIV><B>Sent:</B> Wednesday, October 31, 2012 4:46 PM</DIV>
<DIV><B>To:</B> <A title=asteriskbrasil@listas.asteriskbrasil.org
href="mailto:asteriskbrasil@listas.asteriskbrasil.org"
moz-do-not-send="true">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">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">sip:97095313@189.47.46.151:5060</A><BR>Contact: <A
class=moz-txt-link-rfc2396E href="sip:100@189.2.20.134"
moz-do-not-send="true">sip:100@189.2.20.134</A><BR>Call-ID: <A
href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">sip:97095313@189.47.46.151:5060</A><BR><A
class=moz-txt-link-abbreviated
href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">sip:100@189.2.20.134</A><BR>Call-ID: <A
href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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
href="mailto:%27644015366b7e382f266705505a200615@189.2.20.134%27"
moz-do-not-send="true">mailto:%27644015366b7e382f266705505a200615@189.2.20.134%27</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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">sip:100@189.2.20.134</A>;tag=as57399d63<BR>Call-ID:
<A href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134"
moz-do-not-send="true">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
href="mailto:%273abbce8a451fb58b6726e36623e98a89@189.2.20.134%27"
moz-do-not-send="true">mailto:%273abbce8a451fb58b6726e36623e98a89@189.2.20.134%27</A>
Method: BYE<BR>[Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really destroying
SIP dialog <A href="mailto:%27CE40-3FA8-4674241856E1641BE1B1-193@SipHost%27"
moz-do-not-send="true">mailto:%27CE40-3FA8-4674241856E1641BE1B1-193@SipHost%27</A>
Method: REGISTER<BR>[Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really
destroying SIP dialog <A
href="mailto:%27CE40-3FA8-4674242047F2EB3729E8-194@SipHost%27"
moz-do-not-send="true">mailto:%27CE40-3FA8-4674242047F2EB3729E8-194@SipHost%27</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 title=ivan.paes@gmail.com
href="mailto:ivan.paes@gmail.com" moz-do-not-send="true">Ivan Paes José</A>
</DIV>
<DIV><B>Sent:</B> Tuesday, October 30, 2012 12:20 AM</DIV>
<DIV><B>To:</B> <A title=asteriskbrasil@listas.asteriskbrasil.org
href="mailto:asteriskbrasil@listas.asteriskbrasil.org"
moz-do-not-send="true">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
href="mailto:ivan.paes@gmail.com" target=_blank
moz-do-not-send="true">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 href="mailto:ivanorosco@hotmail.com" target=_blank
moz-do-not-send="true">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 href="http://189.2.20.138:19292"
target=_blank moz-do-not-send="true">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 href="http://189.2.20.138:19292" target=_blank
moz-do-not-send="true">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 href="http://www.khomp.com"
target=_blank
moz-do-not-send="true">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 href="http://www.digivoice.com.br"
target=_blank
moz-do-not-send="true">www.digivoice.com.br</A><BR>________<BR>YEALINK:
Telefones IP e VídeoPhones IP com o melhor custo/benefício do
mercado.<BR>email: <A href="mailto:yealink@commlogik.com.br"
moz-do-not-send="true">yealink@commlogik.com.br</A> | <A
href="http://www.commlogik.com.br" target=_blank
moz-do-not-send="true">www.commlogik.com.br</A> | <A
href="tel:%2811%29%205503-1011" moz-do-not-send="true"
value="+551155031011">(11)
5503-1011</A><BR>______________________________________________<BR>Para
remover seu email desta lista, basta enviar um email em branco para <A
href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org"
moz-do-not-send="true">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</A><BR></BLOCKQUOTE></DIV><BR>
<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"
moz-do-not-send="true">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"
moz-do-not-send="true">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"
moz-do-not-send="true">yealink@commlogik.com.br</A> | <A
class=moz-txt-link-abbreviated href="http://www.commlogik.com.br"
moz-do-not-send="true">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"
moz-do-not-send="true">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" moz-do-not-send="true">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" moz-do-not-send="true">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" moz-do-not-send="true">yealink@commlogik.com.br</A> | <A class=moz-txt-link-abbreviated href="http://www.commlogik.com.br" moz-do-not-send="true">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" moz-do-not-send="true">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</A></PRE></BLOCKQUOTE><BR>
<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>
<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
www.khomp.com<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 www.digivoice.com.br<BR>________<BR>YEALINK:
Telefones IP e VídeoPhones IP com o melhor custo/benefício do mercado.<BR>email:
yealink@commlogik.com.br | www.commlogik.com.br | (11)
5503-1011<BR>______________________________________________<BR>Para remover seu
email desta lista, basta enviar um email em branco para
asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</DIV></DIV></DIV></BODY></HTML>