<br><br>---------- Forwarded message ----------<br><span class="gmail_quote">From: <b class="gmail_sendername">Vidura Senadeera</b> <<a href="mailto:vidurased@gmail.com">vidurased@gmail.com</a>><br>Date: Mar 8, 2007 11:27 AM
<br>Subject: Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br><br></span>
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<div>Hi steve and All,</div>
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<div>I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information</div>
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<div>Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how.</div>
<div>To day also reported 10 call drops within 2 hours of period.</div>
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<div>fook forward to have your support on this regard.</div>
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<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Thanks & Regards,</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Vidura Senadeera,</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Network Engineer,</font></span></p></div>
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<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Debug Solutions</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Sri Lanka.</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Tel - +94114520036</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Mobile - +94777766596</font></span></p>
<p style="MARGIN: 0in 0in 0pt"><span style="COLOR: #333399"><font face="Times New Roman">Web - <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.debug.lk/" target="_blank">www.debug.lk</a></font>
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<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Message: 16<br>Date: Wed, 7 Mar 2007 05:05:36 -0500<br>From: "Steve Totaro" <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:stotaro@asteriskhelpdesk.com" target="_blank">
stotaro@asteriskhelpdesk.com</a>><br>Subject: RE: [asterisk-users] Back to back E1 - asterisk <=> toshiba<br> pbx - Calldroping issue<br>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>Message-ID:<br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:DFB93BD730105941BD1A782A1EE9E95CCC76@1-0fa9e300af524.asteriskhelpdesk.com" target="_blank">
DFB93BD730105941BD1A782A1EE9E95CCC76@1-0fa9e300af524.asteriskhelpdesk.com</a>><br><br>Content-Type: text/plain; charset="us-ascii"<br><br>As these problems are very time sensitive and frustrating, I suggest you
<br>document each change you make and do them one at a time so you can<br>actually know what the problem was and not introduce new problems in the<br>process.<br><br><br><br>Find someone who is on the phone quite a bit and will give you an honest
<br>evaluation of the call dropping situation (unless you yourself are<br>experiencing this issue too). Some people are so quick to say, "It is<br>still happening" without starting the evaluation from a clean slate
<br>after each change.<br><br><br><br>You may want to check your Asterisk log for more insight.<br>/var/log/asterisk/full. Also you can turn on debugging on one span at a<br>time and see if you can find something there<br>
<br><br><br>Do you have a resetinterval set in zapata.conf? If you can isolate the<br>dropped calls to the reset interval (watch the console, it will scroll<br>with each channel being reset) then set resetinterval=never. If there
<br>is no entry for resetinterval, add it and set it to never since it is<br>defaulted to on.<br><br><br><br>Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4<br>to span=2,0,0,ccs,hdb3,crc4. This in combination with your first span
<br>should accept timing from the Telco and then supply it to your Toshiba,<br>I would actually try this first.<br><br><br><br>Another thought, It seems you have quite a lot of hardware in that box.<br>I am not sure how much is too much, but that would probably just rear
<br>it's ugly head as poor audio.<br><br>Thanks,<br>Steve Totaro<br><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.asteriskhelpdesk.com/" target="_blank">http://www.asteriskhelpdesk.com</a>
<br><br><br>_____<br><br>From: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a><br>[mailto:<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">
asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Vidura<br>Senadeera<br>Sent: Wednesday, March 07, 2007 2:15 AM <br>To: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:support@digium.com" target="_blank">
support@digium.com</a><br>Cc: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx -
<br>Calldroping issue<br><br><br><br><br><br>Hi Team,<br><br><br><br>I have integrated asterisk with Toshiba analog PBX. NOw the live setup<br>is going.<br><br><br><br>Now I am facing call droping problem. It's happening ample time. 10-20
<br>calls are droping every day.<br><br><br><br>What could be the reason. I attached latest zapata.conf file for your<br>information.<br><br><br><br><br><br><br><br>This is being a huge issue.<br><br><br><br>Highly appreciate your help on this regard.
<br><br><br>Thanks & Regards,<br><br>Vidura Senadeera.<br><br><br><br><br>On 1/26/07, Vidura Senadeera <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:vidurased@gmail.com" target="_blank">vidurased@gmail.com
</a> > wrote:<br><br>Dear Marco,<br><br><br><br>There is a huge problem i'm facing. <br><br><br><br>My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i<br>conected to the telco. other E1 port i'm using to cros-connection with
<br>toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx <br>E1 not getting. d-channels are not getting up.<br><br>what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12.<br><br><br><br>
notes - if i put, zap show channels in asterisk cli. its only showing<br>the first 31 channels. but with ztcfg -vvv it showing al the channels.<br><br><br><br>my configs are<br><br><br><br># Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED
<br><br># ============ Suntel E1 connection ========== <br><br>span=1,1,0,ccs,hdb3,crc4<br>bchan=1-15,17-31<br>dchan=16<br><br># Span 2: WCT1/1 "Digium Wildcard TE110P T1/E1 Card 1"<br># ============ Legacy PBX E1 connection =======
<br><br>span=2,2,0,ccs,hdb3,crc4<br>bchan=32-46,48-62<br>dchan=47<br><br># Span 3: WCTDM/0 "Wildcard TDM2400P Prototype Board 1"<br>fxoks=63<br>fxoks=64<br>fxoks=65<br>fxoks=66<br>fxoks=67<br>fxoks=68<br>fxoks=69
<br>fxoks=70<br>fxoks=71<br>fxoks=72 <br>fxoks=73<br>fxoks=74<br>fxoks=75<br>fxoks=76<br>fxoks=77<br>fxoks=78<br>fxoks=79<br>fxoks=80<br>fxoks=81<br>fxoks=82<br>fxsks=83<br>fxsks=84<br>fxsks=85<br>fxsks=86<br><br># Global data
<br><br>loadzone = us<br>defaultzone = us<br><br>Regards,<br><br>vidura<br><br><br><br><br><br>--<br>Thanks & Regards,<br>Vidura B. Senadeera.<br><br><br><br><br>--<br>Thanks & Regards,<br>Vidura B. Senadeera.
<br><br><br><br><br>--<br>Thanks & Regards,<br>Vidura B. Senadeera.<br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62a82c69/attachment-0001.htm" target="_blank">
http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62a82c69/attachment-0001.htm</a><br><br>------------------------------<br><br>Message: 17<br>Date: Wed, 7 Mar 2007 11:17:07 +0100<br>From: "Thomas Deillon" <
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:Thomas.Deillon@smart-telecom.ch" target="_blank">Thomas.Deillon@smart-telecom.ch</a>><br>Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem
<br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com
</a>><br>Message-ID:<br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:86918CDC1242004D8B0563A43D1E2F0C027E2D09@exch-pul-01.interne.smart-telecom.ch" target="_blank"> 86918CDC1242004D8B0563A43D1E2F0C027E2D09@exch-pul-01.interne.smart-telecom.ch
</a>><br><br>Content-Type: text/plain; charset="us-ascii"<br><br>Hi all,<br><br><br><br>I install the Asterisk 1.4.1 in order to use the T.38 pass-through, but<br>for the moment, I cannot even make call ....<br>
<br>I have this WARNING:<br><br><br><br>[Mar 7 11:32:09] WARNING[13395]: chan_sip.c:12290 handle_response:<br>Remote host can't match request BYE to call <br>'<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:5759b80c119e1d51679dc66b519c6eac@194.148.41.50" target="_blank">
5759b80c119e1d51679dc66b519c6eac@194.148.41.50</a>'. Giving up.<br><br><br><br>Do you know what is this error and what can I do to solve it ?<br><br><br><br>Thanks a lot for your help,<br><br><br><br>Thomas<br><br><br>
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<br>-- <br>Thanks & Regards,<br>Vidura B. Senadeera. </span><br clear="all"><br>-- <br>Thanks & Regards,<br>Vidura B. Senadeera.