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

Tony Nichols tony.nichols at gmail.com
Wed Dec 3 14:54:21 CST 2008


On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

>
>
> 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)
>
> _______________________________________________
> -- 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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