[asterisk-users] * + Legacy PBX works but strange problem

[Steve Totaro]

On Sun, Nov 16, 2008 at 4:28 AM, Sriram <d_r_sriram at hotmail.com> 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)
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