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<body class='hmmessage'><div dir='ltr'>Não não, meu caro, o meu provedor requer que receba o 55, mas não é para ligação internacional, é apenas a forma que estão as rotas dele.<div><br></div><div>Era problema no codec do meu tronco mesmo, meu ramal estava para alaw, porem o tronco apenas aceitava g729. ^^</div><div><br></div><div>Grato!<br><br><div><hr id="stopSpelling">Date: Tue, 1 Oct 2013 11:11:16 -0300<br>From: biro0702@gmail.com<br>To: asteriskbrasil@listas.asteriskbrasil.org<br>Subject: Re: [AsteriskBrasil] Magnus Billing<br><br><div dir="ltr">Não esta faltando um "00" antes do número "<span style="font-family:arial,sans-serif;font-size:13px;">556492361315", ou então retirar o "55", e enviar apenas o "0" antes do "</span><span style="font-family:arial,sans-serif;font-size:13px;">6492361315" ?</span></div>
<div class="ecxgmail_extra"><br><br><div class="ecxgmail_quote">On Tue, Oct 1, 2013 at 11:01 AM, Wilson Ritt Iglesias <span dir="ltr"><<a href="mailto:wilson.ritt@hotmail.com" target="_blank">wilson.ritt@hotmail.com</a>></span> wrote:<br>
<blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">
<div><div dir="ltr"><div><div style="display:inline-block;"><span>Não possuo g729 no Elastix.</span></div></div><div><br></div><div>Desabilitei conforme imagem abaixo;</div><div><div style="display:inline-block;"><span><br>
</span></div></div><div style="display:inline-block;"><span> </span><div style="display:inline-block;"><img height="387" width="399"></div><span> </span></div><div><br></div><div><div style="display:inline-block;"><span> </span><div style="display:inline-block;">
<img src="cid:inlineImage1" height="707" width="548"></div><span> </span></div><br></div><div><br></div><div>Ainda com o mesmo problema =\</div><div><br></div><div><div class="ecxim"><div>localhost*CLI></div><div> == Using SIP RTP CoS mark 5</div>
</div><div> -- Executing [556492361315@billing:1] AGI("SIP/3000-00000064", "magnus")</div><div class="ecxim"><div> -- Launched AGI Script /var/lib/asterisk/agi-bin/magnus</div><div> -- AGI Script Executing Application: (DIAL) Options: (sip/Brasiltel/556492361315,60,L(300000000:61000:30000))</div>
</div><div> > Limit Data for this call:</div><div> > timelimit = 300000000 ms (300000.000 s)</div><div> > play_warning = 61000 ms (61.000 s)</div><div> > play_to_caller = yes</div>
<div> > play_to_callee = no</div><div> > warning_freq = 30000 ms (30.000 s)</div><div> > start_sound =</div><div> > warning_sound = timeleft</div><div> > end_sound =</div>
<div class="ecxim"><div> == Using SIP RTP CoS mark 5</div></div><div>[Oct 1 09:57:22] WARNING[4243]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315</div><div class="ecxim"><div> -- Couldn't call sip/Brasiltel/556492361315</div>
<div> == Everyone is busy/congested at this time (0:0/0/0)</div></div><div> -- <SIP/3000-00000064>AGI Script magnus completed, returning 0</div><div> -- Executing [556492361315@billing:2] Hangup("SIP/3000-00000064", "")</div>
<div> == Spawn extension (billing, 556492361315, 2) exited non-zero on 'SIP/3000-00000064'</div><div><br></div><br><div><hr>From: <a href="mailto:info@magnussolution.com" target="_blank">info@magnussolution.com</a><br>
Date: Tue, 1 Oct 2013 10:52:29 -0300<div><div class="h5"><br>To: <a href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>Subject: Re: [AsteriskBrasil] Magnus Billing<br>
<br>ola, seu Elastix e o Mbilling esta com codec 729?<div><br></div><div>Se nao tiver, desative estes codec no tronco do Mbilling</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br><div><div>On oct 1, 2013, at 10:42 a.m., Wilson Ritt Iglesias <<a href="mailto:wilson.ritt@hotmail.com" target="_blank">wilson.ritt@hotmail.com</a>> wrote:</div>
<br><blockquote><div style="font-size:12pt;font-family:Calibri;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:-webkit-auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;">
<div dir="ltr">Perdão, peguei a saída do Elastix e não no MBilling.<div><br></div><div>Continuo tendo o mesmo problema:</div><div><br></div><div><div>localhost*CLI></div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [556492361315@billing:1] AGI("SIP/3000-00000058", "magnus")</div>
<div> -- Launched AGI Script /var/lib/asterisk/agi-bin/magnus</div><div> -- AGI Script Executing Application: (DIAL) Options: (sip/Brasiltel/556492361315,60,L(300000000:61000:30000))</div><div> == Using SIP RTP CoS mark 5</div>
<div>[Oct 1 09:41:54] WARNING[4032]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315</div><div> -- Couldn't call sip/Brasiltel/556492361315</div><div> == Everyone is busy/congested at this time (0:0/0/0)</div>
<div> -- <SIP/3000-00000058>AGI Script magnus completed, returning 0</div><div> -- Executing [556492361315@billing:2] Hangup("SIP/3000-00000058", "")</div><div> == Spawn extension (billing, 556492361315, 2) exited non-zero on 'SIP/3000-00000058'</div>
<div><br></div><br><div><hr>From:<span> </span><a href="mailto:wilson.ritt@hotmail.com" target="_blank">wilson.ritt@hotmail.com</a><br>To:<span> </span><a href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>
Date: Tue, 1 Oct 2013 10:17:40 -0300<br>Subject: Re: [AsteriskBrasil] Magnus Billing<br><br><div dir="ltr">Liberei todos os codecs no MBilling, e alterou a saída quando tento realizar a ligação, porém ainda recebo tom de ocupado... Pelo visto agora deu algum declined...<div>
(<span style="font-size:12pt;">-- Got SIP response 603 "Declined" back from <a href="http://189.38.32.8:5060" target="_blank">189.38.32.8:5060</a>)</span></div><div><span style="font-size:12pt;"><br></span></div><div>
<div>pabx*CLI></div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [556492361315@from-internal:1] Macro("SIP/5236-00006183", "user-callerid,SKIPTTL,") in new stack</div>
<div> -- Executing [s@macro-user-callerid:1] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:2] GotoIf("SIP/5236-00006183", "0?report") in new stack</div>
<div> -- Executing [s@macro-user-callerid:3] ExecIf("SIP/5236-00006183", "1?Set(REALCALLERIDNUM=5236)") in new stack</div><div> -- Executing [s@macro-user-callerid:4] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack</div>
<div> -- Executing [s@macro-user-callerid:5] Set("SIP/5236-00006183", "AMPUSERCIDNAME=Wilson") in new stack</div><div> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/5236-00006183", "0?report") in new stack</div>
<div> -- Executing [s@macro-user-callerid:7] Set("SIP/5236-00006183", "AMPUSERCID=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:8] Set("SIP/5236-00006183", "CALLERID(all)="Wilson" <5236>") in new stack</div>
<div> -- Executing [s@macro-user-callerid:9] ExecIf("SIP/5236-00006183", "0?Set(CHANNEL(language)=)") in new stack</div><div> -- Executing [s@macro-user-callerid:10] GotoIf("SIP/5236-00006183", "1?continue") in new stack</div>
<div> -- Goto (macro-user-callerid,s,19)</div><div> -- Executing [s@macro-user-callerid:19] Set("SIP/5236-00006183", "CALLERID(number)=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:20] Set("SIP/5236-00006183", "CALLERID(name)=Wilson") in new stack</div>
<div> -- Executing [s@macro-user-callerid:21] NoOp("SIP/5236-00006183", "Using CallerID "Wilson" <5236>") in new stack</div><div> -- Executing [556492361315@from-internal:2] NoOp("SIP/5236-00006183", "Calling Out Route: Magnus_Teste") in new stack</div>
<div> -- Executing [556492361315@from-internal:3] Set("SIP/5236-00006183", "MOHCLASS=default") in new stack</div><div> -- Executing [556492361315@from-internal:4] Set("SIP/5236-00006183", "_NODEST=") in new stack</div>
<div> -- Executing [556492361315@from-internal:5] Macro("SIP/5236-00006183", "record-enable,5236,OUT,") in new stack</div><div> -- Executing [s@macro-record-enable:1] GotoIf("SIP/5236-00006183", "1?check") in new stack</div>
<div> -- Goto (macro-record-enable,s,4)</div><div> -- Executing [s@macro-record-enable:4] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack</div><div> -- Executing [s@macro-record-enable:5] GotoIf("SIP/5236-00006183", "0?Group:OUT") in new stack</div>
<div> -- Goto (macro-record-enable,s,15)</div><div> -- Executing [s@macro-record-enable:15] GotoIf("SIP/5236-00006183", "0?IN") in new stack</div><div> -- Executing [s@macro-record-enable:16] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack</div>
<div> -- Executing [s@macro-record-enable:17] NoOp("SIP/5236-00006183", "Recording enable for 5236") in new stack</div><div> -- Executing [s@macro-record-enable:18] Set("SIP/5236-00006183", "CALLFILENAME=OUT5236-20131001-131356-1380644036.77600") in new stack</div>
<div> -- Executing [s@macro-record-enable:19] Goto("SIP/5236-00006183", "record") in new stack</div><div> -- Goto (macro-record-enable,s,23)</div><div> -- Executing [s@macro-record-enable:23] MixMonitor("SIP/5236-00006183", "OUT5236-20131001-131356-1380644036.77600.gsm,,") in new stack</div>
<div> -- Executing [s@macro-record-enable:24] Set("SIP/5236-00006183", "CDR(userfield)=audio:OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div><div> -- Executing [s@macro-record-enable:25] MacroExit("SIP/5236-00006183", "") in new stack</div>
<div> -- Executing [556492361315@from-internal:6] Macro("SIP/5236-00006183", "dialout-trunk,1,556492361315,") in new stack</div><div> -- Executing [s@macro-dialout-trunk:1] Set("SIP/5236-00006183", "DIAL_TRUNK=1") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/5236-00006183", "0?sub-pincheck,s,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/5236-00006183", "0?disabletrunk,1") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:4] Set("SIP/5236-00006183", "DIAL_NUMBER=556492361315") in new stack</div><div> -- Executing [s@macro-dialout-trunk:5] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=tr") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:6] Set("SIP/5236-00006183", "OUTBOUND_GROUP=OUT_1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/5236-00006183", "1?nomax") in new stack</div>
<div> -- Goto (macro-dialout-trunk,s,9)</div><div> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/5236-00006183", "0?skipoutcid") in new stack</div><div> -- Executing [s@macro-dialout-trunk:10] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/5236-00006183", "outbound-callerid,1") in new stack</div><div> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=)") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/5236-00006183", "0?Set(REALCALLERIDNUM=5236)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/5236-00006183", "1?normcid") in new stack</div>
<div> -- Goto (macro-outbound-callerid,s,6)</div><div> -- Executing [s@macro-outbound-callerid:6] Set("SIP/5236-00006183", "USEROUTCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:7] Set("SIP/5236-00006183", "EMERGENCYCID=") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:8] Set("SIP/5236-00006183", "TRUNKOUTCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/5236-00006183", "1?trunkcid") in new stack</div>
<div> -- Goto (macro-outbound-callerid,s,12)</div><div> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/5236-00006183", "0?sub-flp-1,s,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:13] Set("SIP/5236-00006183", "OUTNUM=556492361315") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:14] Set("SIP/5236-00006183", "custom=SIP/Magnus_Billing") in new stack</div><div> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/5236-00006183", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:16] Macro("SIP/5236-00006183", "dialout-trunk-predial-hook,") in new stack</div><div> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/5236-00006183", "") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/5236-00006183", "0?bypass,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/5236-00006183", "0?customtrunk") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:19] Dial("SIP/5236-00006183", "SIP/Magnus_Billing/556492361315,300,") in new stack</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div>
<div> -- Called SIP/Magnus_Billing/556492361315</div><div> == Begin MixMonitor Recording SIP/5236-00006183</div><div> -- Got SIP response 603 "Declined" back from <a href="http://189.38.32.8:5060" target="_blank">189.38.32.8:5060</a></div>
<div> -- SIP/Magnus_Billing-00006184 is busy</div><div> == Everyone is busy/congested at this time (1:1/0/0)</div><div> -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/5236-00006183", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:21] Goto("SIP/5236-00006183", "s-BUSY,1") in new stack</div><div> -- Goto (macro-dialout-trunk,s-BUSY,1)</div><div> -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/5236-00006183", "Dial failed due to trunk reporting BUSY - giving up") in new stack</div>
<div> -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/5236-00006183", "busy") in new stack</div><div> -- Executing [s-BUSY@macro-dialout-trunk:3] Busy("SIP/5236-00006183", "20") in new stack</div>
<div> == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on 'SIP/5236-00006183' in macro 'dialout-trunk'</div><div> == Spawn extension (from-internal, 556492361315, 6) exited non-zero on 'SIP/5236-00006183'</div>
<div> -- Executing [h@from-internal:1] Macro("SIP/5236-00006183", "hangupcall") in new stack</div><div> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack</div>
<div> -- Executing [s@macro-hangupcall:2] Set("SIP/5236-00006183", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div><div> -- Executing [s@macro-hangupcall:3] GotoIf("SIP/5236-00006183", "1?defaultmixmondir") in new stack</div>
<div> -- Goto (macro-hangupcall,s,5)</div><div> -- Executing [s@macro-hangupcall:5] System("SIP/5236-00006183", "test -e /var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div>
<div> -- Executing [s@macro-hangupcall:6] NoOp("SIP/5236-00006183", "SYSTEMSTATUS = APPERROR") in new stack</div><div> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack</div>
<div> -- Executing [s@macro-hangupcall:8] Set("SIP/5236-00006183", "CDR(userfield)=") in new stack</div><div> -- Executing [s@macro-hangupcall:9] NoOp("SIP/5236-00006183", "End of MIXMON check") in new stack</div>
<div> -- Executing [s@macro-hangupcall:10] GotoIf("SIP/5236-00006183", "1?nomeetmemon") in new stack</div><div> -- Goto (macro-hangupcall,s,28)</div><div> -- Executing [s@macro-hangupcall:28] NoOp("SIP/5236-00006183", "End of MEETME check") in new stack</div>
<div> -- Executing [s@macro-hangupcall:29] GotoIf("SIP/5236-00006183", "1?noautomon") in new stack</div><div> -- Goto (macro-hangupcall,s,34)</div><div> -- Executing [s@macro-hangupcall:34] NoOp("SIP/5236-00006183", "TOUCH_MONITOR_OUTPUT=") in new stack</div>
<div> -- Executing [s@macro-hangupcall:35] GotoIf("SIP/5236-00006183", "1?noautomon2") in new stack</div><div> -- Goto (macro-hangupcall,s,41)</div><div> -- Executing [s@macro-hangupcall:41] NoOp("SIP/5236-00006183", "MONITOR_FILENAME=") in new stack</div>
<div> -- Executing [s@macro-hangupcall:42] GotoIf("SIP/5236-00006183", "1?skiprg") in new stack</div><div> -- Goto (macro-hangupcall,s,45)</div><div> -- Executing [s@macro-hangupcall:45] GotoIf("SIP/5236-00006183", "1?skipblkvm") in new stack</div>
<div> -- Goto (macro-hangupcall,s,48)</div><div> -- Executing [s@macro-hangupcall:48] GotoIf("SIP/5236-00006183", "1?theend") in new stack</div><div> -- Goto (macro-hangupcall,s,50)</div><div>
-- Executing [s@macro-hangupcall:50] Hangup("SIP/5236-00006183", "") in new stack</div><div> == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/5236-00006183' in macro 'hangupcall'</div>
<div> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5236-00006183'</div><div> == End MixMonitor Recording SIP/5236-00006183</div><div><br></div><div><br></div><br><div><hr>Date: Tue, 1 Oct 2013 10:13:05 -0300<br>
From: <a href="mailto:giovbs@gmail.com" target="_blank">giovbs@gmail.com</a><br>To: <a href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>Subject: Re: [AsteriskBrasil] Magnus Billing<br>
<br><div dir="ltr">Wilson,<div><br></div><div>verifique os codecs pois está retornando este erro:<br><br><font color="#191919" face="Arial, Helvetica, sans-serif"><span style="font-size:14px;line-height:22.171875px;">No audio format found to offer. Cancelling call to </span></font><a style="font-family:Arial,Helvetica,sans-serif;font-size:14px;line-height:22.171875px;" target="_blank">556492361315</a><br>
<div><br>Abraço.<br><br><br><div>Em 1 de outubro de 2013 10:07, Wilson Ritt Iglesias<span> </span><span dir="ltr"><<a href="mailto:wilson.ritt@hotmail.com" target="_blank">wilson.ritt@hotmail.com</a>></span><span> </span>escreveu:<br>
<blockquote style="border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex;"><div dir="ltr"><div style="line-height:21px;color:rgb(68,68,68);font-size:15px;padding:10px 0px 0px;display:table;table-layout:fixed;width:1235px;min-height:180px;">
<div style="line-height:22.171875px;color:rgb(25,25,25);font-family:Arial,Helvetica,sans-serif;font-size:0.95em;padding:0px;word-wrap:break-word;overflow:hidden;">Ao tentar ligar, tenho essas saídas no painel do asterisk:</div>
<div style="line-height:22.171875px;color:rgb(25,25,25);font-family:Arial,Helvetica,sans-serif;font-size:0.95em;padding:0px;word-wrap:break-word;overflow:hidden;"><br style="line-height:19px;"></div><div style="line-height:22.171875px;color:rgb(25,25,25);font-family:Arial,Helvetica,sans-serif;font-size:0.95em;padding:0px;word-wrap:break-word;overflow:hidden;">
Ligando pelo tronco no Elastix que criei:<br style="line-height:19px;padding:0px;"><br><br><br></div></div></div></blockquote><blockquote style="border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex;">
<div dir="ltr"><div style="line-height:21px;color:rgb(68,68,68);font-size:15px;padding:10px 0px 0px;display:table;table-layout:fixed;width:1235px;min-height:180px;"><div style="line-height:22.171875px;color:rgb(25,25,25);font-family:Arial,Helvetica,sans-serif;font-size:0.95em;padding:0px;word-wrap:break-word;overflow:hidden;">
<span style="font-size:0.95em;">[Oct 1 08:46:58] WARNING[3307]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to<span> </span></span><a style="font-size:0.95em;" target="_blank">556492361315</a><br>-- Couldn't call sip/Brasiltel/<a target="_blank">556492361315</a><br style="line-height:19px;padding:0px;">
</div></div></div></blockquote></div></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><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<span> </span><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<span> </span><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<span> </span><a href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org" target="_blank">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div>
</div></div>_______________________________________________<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></div>
</blockquote></div><br></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">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br></blockquote></div>
<br></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 www.Khomp.com.
_______________________________________________
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 www.aligera.com.br.
_______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</div></div> </div></body>
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