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<p class=MsoNormal><font size=3 color=navy face=Arial><span style='font-size:
12.0pt;font-family:Arial;color:navy'>Before studying your configs, what have
you tried so far?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color=navy face=Arial><span style='font-size:
12.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color=navy face=Arial><span style='font-size:
12.0pt;font-family:Arial;color:navy'>Did you change this? <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color=navy face=Arial><span style='font-size:
12.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'>Also, try changing your second span timing from
span=2,2,0,ccs,hdb3,crc4<br>
to span=2,0,0,ccs,hdb3,crc4.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'>Here is the documentation on voip-info for why it may be the
cause of your issues<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'>http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face=Arial><span style='font-size:12.0pt;
font-family:Arial'><br>
span definition format: <br>
span=(spannum),(timing),(LBO),(framing),(coding) <br>
<br>
spannum= Number of the span. <br>
<br>
timing= How to synchronize the timing devices. <br>
0: to not use this span as sync source <br>
1: to use as primary sync source <br>
2: to set as secondary and so forth <br>
<br>
Use '1' if you want to use the circuit as your primary sync source. If '0' is
used asterisk will try to provide timing to the span (say, if you were
connecting to a legacy PBX). If Asterisk is connected directly to the telco you
will want to use '1' to accept timing from them. If youhave multiple spans, set
them as 2, 3, 4, etc. <br>
<br>
Problems with timing manifest themselves different ways - with static, pops,
and channels or calls regularly dropping.<font color=navy><span
style='color:navy'><o:p></o:p></span></font></span></font></p>
<p class=MsoNormal><font size=3 color=navy face=Arial><span style='font-size:
12.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<p><font size=3 color=navy face=Arial><span style='font-size:12.0pt;font-family:
Arial;color:navy'>Thanks,<br>
Steve Totaro<br>
<a href="http://www.asteriskhelpdesk.com">http://www.asteriskhelpdesk.com</a><br>
KB3OPB<br>
</span></font><font face=Arial><span style='font-family:Arial'><o:p></o:p></span></font></p>
</div>
<div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'>
<div>
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face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Vidura Senadeera<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, March 08, 2007
1:01 AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Fwd:
Back to back E1 - asterisk <=> toshiba pbx -Call droping</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'><br>
<br>
---------- Forwarded message ----------<br>
<span class=gmailquote>From: <b><span style='font-weight:bold'>Vidura Senadeera</span></b>
<<a href="mailto:vidurased@gmail.com">vidurased@gmail.com</a>></span><br>
<span class=gmailquote>Date: Mar 8, 2007 11:27 AM </span><br>
<span class=gmailquote>Subject: Re: Back to back E1 - asterisk <=>
toshiba pbx - Call droping</span><br>
<span class=gmailquote>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></span><o:p></o:p></span></font></p>
<div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi steve and All,<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I'm attaching cat proc/interrupts out put, lspci -bv out put,
zapa.conf, zaptel.conf for your information<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Thanks so much for the feedback and I do accordingly. Hope to get rid
off this isue any how.<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>To day also reported 10 call drops within 2 hours of period.<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>fook forward to have your support on this regard.<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Thanks
& Regards,</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Vidura
Senadeera,</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Network
Engineer,</span></font><o:p></o:p></p>
</div>
<div>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Debug
Solutions</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><st1:country-region w:st="on"><st1:place
w:st="on"><font size=3 color="#333399" face="Times New Roman"><span
style='font-size:12.0pt;color:#333399'>Sri Lanka</span></font></st1:place></st1:country-region><font
color="#333399"><span style='color:#333399'>.</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Tel -
+94114520036</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><st1:City w:st="on"><st1:place
w:st="on"><font size=3 color="#333399" face="Times New Roman"><span
style='font-size:12.0pt;color:#333399'>Mobile</span></font></st1:place></st1:City><font
color="#333399"><span style='color:#333399'> - +94777766596</span></font><o:p></o:p></p>
<p style='margin:0in;margin-bottom:.0001pt'><font size=3 color="#333399"
face="Times New Roman"><span style='font-size:12.0pt;color:#333399'>Web - <a
href="http://www.debug.lk/" target="_blank">www.debug.lk</a> </span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Message: 16<br>
Date: Wed, 7 Mar 2007 05:05:36 -0500<br>
From: "Steve Totaro" <<a 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: "<st1:PersonName w:st="on">Asterisk Users Mailing List -
Non-Commercial Discussion</st1:PersonName>" <br>
<<a
href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<<a
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 href="http://www.asteriskhelpdesk.com/" target="_blank">http://www.asteriskhelpdesk.com</a>
<br>
<br>
<br>
_____<br>
<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a 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 href="mailto:support@digium.com" target="_blank">support@digium.com</a><br>
Cc: <a 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 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>
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<br>
------------------------------<br>
<br>
Message: 17<br>
Date: Wed, 7 Mar 2007 11:17:07 +0100<br>
From: "Thomas Deillon" < <a
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: "<st1:PersonName w:st="on">Asterisk Users Mailing List -
Non-Commercial Discussion</st1:PersonName>" <br>
<<a
href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com
</a>><br>
Message-ID:<br>
<<a
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 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>
<br>
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**********************************************<o:p></o:p></span></font></p>
</blockquote>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<br clear=all>
<br>
<span class=sg>-- </span><br>
<span class=sg>Thanks & Regards,</span><br>
<span class=sg>Vidura B. Senadeera. </span><br clear=all>
<br>
-- <br>
Thanks & Regards,<br>
Vidura B. Senadeera. <o:p></o:p></span></font></p>
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