<br><br><div class="gmail_quote">On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br><br><div class="gmail_quote"><div><div></div><div class="Wj3C7c">On Sun, Nov 16, 2008 at 4:28 AM, Sriram <span dir="ltr"><<a href="mailto:d_r_sriram@hotmail.com" target="_blank">d_r_sriram@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff">
<div>
<div><font face="Arial" size="2"></font><font face="Arial" size="2"></font> </div>
<div>
<div><font face="Arial" size="2">Hi</font></div>
<div><font face="Arial" size="2">below are my configs:</font></div>
<div><font face="Arial" size="2">pstn(e1)--->asterisk (span1)----->legacy
pbx(connected via span2)-----> legacy pbx analog
extensions.</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">my dial plan is like callers dial into
asterisk(span1) , hear an IVR option and they are connected to the agents via
the legacy pbx (which is in sync with asterisk on span2)....This works perfectly
fine until about 200 calls or so...After that time when asterisk tries to dial
to the legacy pbx - the call drops with error "All are busy congested at this
time" .the same is indicated on asterisk -rvvvvvvvvvv , but the spans are up and
active at that time... can anyone throw some light on this ?</font></div>
<div><font face="Arial" size="2"></font> </div>
<div>>>> ZAPTEL.CONF <br><pre><code>
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
bchan=32-46
dchan=47
bchan=48-62
>>> ZAPATA.CONF <br></code><pre><code>
context=pri-pstn
switchtype=euroisdn
pridialplan=local
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=1
pickupgroup=1
immediate=yes
musiconhold=default
signalling = pri_cpe
channel => 1-15
channel => 17-31
context=pri-legacy
immediate=yes
group=2
overlapdial=yes
signalling = pri_net
channel => 32-46
channel => 48-62</code></pre><pre><code>>>> EXTENSIONS.CONF <br></code><pre><code>
;
; Context PRI-Public
;
[pri-pstn]
;
include => default
;
exten => s,1,Answer </code></pre><pre><code>exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx
exten => s,3,Hangup
;
; Context PRI-legacy
;
[pri-legacy]
;
include => default
;
exten => s,1,Answer
exten => s,2,DigitTimeout,2
exten => s,3,ResponseTimeout,2
exten => _X.,1,Dial(Zap/g1/${EXTEN})
exten => _X.,2,Congestion</code>
</pre></pre></pre></div></div></div></div>
</blockquote></div></div><div><br>This is just a suggestion that has worked very well for me in the past when dealing with "Legacy" systems that have only "Analog" phones connected.<br><br>Ditch the Legacy system and get some form of channel bank. If you want to go SIP to Analog, I have had great luck with Quintum Tenor AX. Since, you have a spare E1 port, you could simply terminate the analog lines to a tried and true channel bank. I have never looked for an E1 channel bank (30 port density) but I would assume they exist. <br>
<br>If the Legacy system has proprietary, digital extensions, that complicates things a bit. <br><br>Special apps running or connected on your Legacy system can usually be migrated and after that bit of growing pain, you have all the flexibility you want to customize.<br>
</div></div><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>
<br>_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
</blockquote><div><br>I have noticed when connecting our legacy system to asterisk, the option "<pre><pre><code>overlapdial=yes<br><br><br></code></pre></pre> </div></div>caused issues with only certain exchanges... and would appear randomly. It seems to add a "pause" of some 4 sec. when dialing. <br>
This would give you the "busy" error.<br clear="all"><br>-- <br>A.G. (Tony) Nichols<br>I.S. Manager<br>