On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro < [EMAIL PROTECTED]> wrote:
> > > On Sun, Nov 16, 2008 at 4:28 AM, Sriram <[EMAIL PROTECTED]> wrote: > >> >> Hi >> below are my configs: >> pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> >> legacy pbx analog extensions. >> >> 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 ? >> >> >>> ZAPTEL.CONF >> >> >> 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 >> >> >> 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 >> >> >>> EXTENSIONS.CONF >> >> >> ; >> ; Context PRI-Public >> ; >> [pri-pstn] >> ; >> include => default >> ; >> exten => s,1,Answer >> >> 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 >> >> > 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. > > 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. > > If the Legacy system has proprietary, digital extensions, that complicates > things a bit. > > 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. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > I have noticed when connecting our legacy system to asterisk, the option " overlapdial=yes caused issues with only certain exchanges... and would appear randomly. It seems to add a "pause" of some 4 sec. when dialing. This would give you the "busy" error. -- A.G. (Tony) Nichols I.S. Manager
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