<div dir="ltr">Thanks Steve,<br><br>The reset worked, and now I can access the configuration panel.<br><br>Can you give more details on how should I handle the 30 seconds issue? How could I manage the dial plan to answer the call?<br>
Today it works like this:<br><br><br>Call from PSTN comes in the ATA, it picks up the call and hot dial a group number in my asterisk, this group rings several SIP extensions. I pickup a ringing softphone and start the conversation. This converstion then, hangs up at 30 seconds.<br>
<br><br>The other way round:<br>I pick up a softphone extension, have to dial the ATA number (216), it answers automatically and gives me the PSTN tone for dialing, then I have to dial the number. This call also hangs up at 30 seconds.<br>
<br>I still have not been able to activate a one stage dialing with this ZOOM ATA. Yesterday the support from Zoom send me some instruction (attached below) on how to configure it, but I still have not been able to apply it to my asterisk server.<br>
<br>Any instruction would be nice.<br><br>Thanks,<br><br>Felipe<br><br><br>---------------------------------------Instruction from Zoom Support-------------------------------------------<br>Felipe,<br> <br>This is information on how single stage dialing works in regards to ATA and Asterisk<br>
Enable this when you enable VOIP to PSTN bridging.<br> <br>Enable Single Stage dialing in ATA in the Voip to PSTN Bridging. You also need to setup your asterisk to support this and these are the options.<br> <br>This is how single-stage dialing works:<br>
This feature works by examining the username in the From: header of a SIP INVITE. If the username is different from the username of any account on the ATA, the fxo port will go off hook and automatically dial the number in the username of the From: field.<br>
If the user has configured a security code for VoIP to PSTN dialing, the security code is included as a prefix to the number to dial. If the security code matches, the following digits are dialed out the FXO port. If the security code doesn't match, the call is shunted to the local instrument (i.e. to the FXS port).<br>
Example I: <br>Device is registered as <a href="mailto:6175551212@voipphone.com">6175551212@voipphone.com</a><br>INVITE arrives with From: field 2124442121<br>ATA comes off-hook and dials 2124442121 to the FXO port. It opens a connection between this call and the party that sent the INVITE.<br>
Example II: <br>Device is registered as <a href="mailto:6175551212@voipphone.com">6175551212@voipphone.com</a><br>User has configured security code of 9876<br>INVITE arrives with From: field 98762124442121<br>ATA comes off-hook and dials 2124442121 to the FXO port. It opens a connection between this call and the party that sent the INVITE.<br>
Example III: <br>Device is registered as <a href="mailto:6175551212@voipphone.com">6175551212@voipphone.com</a><br>User has configured security code of 9876<br>INVITE arrives with From: field 67892124442121<br>ATA connects call directly to the FXS port.<br>
Regards<br>ZOom Tech Support<br>Joyce Phillips<br><br>-----------------------End of instructions from Zoom Support-----------------------------------------<br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br>
<br><br><br><br><br><br><br><br><br><div class="gmail_quote">On Thu, Jul 17, 2008 at 6:52 PM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Generally, when you see a call always hang up at 30 seconds it is<br>
because you are not "answering" in your dialplan before doing other<br>
things.<br>
<br>
As for the reset, you may want to hold in the reset button for like 30<br>
seconds, pull the power plug and plug it back in after 10 seconds<br>
while holding down the reset and keep holding it for at least another<br>
30 seconds after you cycle the power.<br>
<br>
Thanks,<br>
Steve T<br>
<div><div></div><div class="Wj3C7c"><br>
On Thu, Jul 17, 2008 at 4:50 PM, Felipe Trevisan <<a href="mailto:asteriskbr99@gmail.com">asteriskbr99@gmail.com</a>> wrote:<br>
> My Zoom 5801 ATA hangs up at 30 seconds every call.<br>
> I do not think it´s an Asterisk issue, as calls on the SIP trunk goes in and<br>
> out normally.<br>
><br>
> Below is the CLI message.<br>
> 216 is the extension number assigned to the FXS extension port on the ATA.<br>
><br>
><br>
> Another problem that came up while I was trying to solve the first problem,<br>
> is that I´ve disabled the internal HTTP server from the ATA, and I can no<br>
> longer access the configuration panel through the browser window. I´ve tried<br>
> a reset puching the small button on the back, but ot simply won´t do<br>
> nothing.<br>
> Any clues?<br>
><br>
> Thanks a lot,<br>
><br>
> Felipe<br>
><br>
><br>
> <------------><br>
><br>
> [Kserver*CLI><br>
> == Spawn extension (macro-dial, s, 7) exited non-zero on<br>
> 'SIP/216-b7803460' in macro 'dial'<br>
> == Spawn extension (macro-dial, s, 7) exited non-zero on<br>
> 'SIP/216-b7803460'<br>
> -- Executing [h@macro-dial:1] [1;36;40mMacro [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40mhangupcall [0;37;40m") in<br>
> new stack<br>
> -- Executing [s@macro-hangupcall:1] [1;36;40mResetCDR [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40mw [0;37;40m") in new stack<br>
> -- Executing [s@macro-hangupcall:2] [1;36;40mNoCDR [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40m [0;37;40m") in new stack<br>
> -- Executing [s@macro-hangupcall:3] [1;36;40mGotoIf [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40m1?skiprg [0;37;40m") in new<br>
> stack<br>
> -- Goto (macro-hangupcall,s,6)<br>
> -- Executing [s@macro-hangupcall:6] [1;36;40mGotoIf [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40m0?skipblkvm [0;37;40m") in<br>
> new stack<br>
> -- Executing [s@macro-hangupcall:7] [1;36;40mNoOp [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40mCleaning Up Block VM Flag:<br>
> BLKVM/601/SIP/216-b7803460 [0;37;40m") in new stack<br>
> -- Executing [s@macro-hangupcall:8] [1;36;40mDBdel [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40mBLKVM/601/SIP/216-<br>
> b7803460 [0;37;40m") in new stack<br>
> -- DBdel: family=BLKVM, key=601/SIP/216-b7803460<br>
> -- DBdel: Error deleting key from database.<br>
> -- Executing [s@macro-hangupcall:9] [1;36;40mGotoIf [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40m1?theend [0;37;40m") in new<br>
> stack<br>
> -- Goto (macro-hangupcall,s,11)<br>
> -- Executing [s@macro-hangupcall:11] [1;36;40mHangup [0;37;40m("<br>
> [1;35;40mSIP/216-b7803460 [0;37;40m", " [1;35;40m [0;37;40m") in new stack<br>
> == Spawn extension (macro-hangupcall, s, 11) exited non-zero on<br>
> 'SIP/216-b7803460' in macro 'hangupcall'<br>
> == Spawn extension (macro-hangupcall, s, 11) exited non-zero on<br>
> 'SIP/216-b7803460'<br>
> Really destroying SIP dialog '<a href="mailto:25d24e06bdb1af06@192.168.0.31">25d24e06bdb1af06@192.168.0.31</a>' Method: BYE<br>
><br>
> [Kserver*CLI><br>
><br>
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