[asterisk-users] Connect Avaya to Asterisk PBX

[Malvin Rito]

How do I write it on my code?

On 7/13/2011 4:04 PM, Warren Selby wrote:
> Looks like you need an 's' exten in your [internal] context.
>
> Thanks,
> --Warren Selby, dCAP
>
> On Jul 13, 2011, at 2:02 AM, Malvin Rito 
> <mrito at mail.altcladding.com.ph <mailto:mrito at mail.altcladding.com.ph>> 
> wrote:
>
>> Hi List,
>>
>> I have another issue on allowing outgoing calls to PSTN on Asterisk 
>> via Avaya Phones, I hope that anyone could help me fix this issue:
>>
>> *When I dial through Avaya phone i just here a "good bye message" 
>> reply from asterisk server. And here is the log:*
>>
>>  == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling 
>> back to exten 's'
>>   == Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so 
>> falling back to context 'default'
>>     -- Executing [s at default:1] Playback("OOH323/(null)-b7db8aa0", 
>> "vm-goodbye") in new stack
>>     -- <OOH323/(null)-b7db8aa0> Playing 'vm-goodbye.ulaw' (language 'en')
>>     -- Executing [s at default:2] Macro("OOH323/(null)-b7db8aa0", 
>> "hangupcall") in new stack
>>     -- Executing [s at macro-hangupcall:1] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?skiprg") in new stack
>>     -- Goto (macro-hangupcall,s,4)
>>     -- Executing [s at macro-hangupcall:4] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?skipblkvm") in new stack
>>     -- Goto (macro-hangupcall,s,7)
>>     -- Executing [s at macro-hangupcall:7] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?theend") in new stack
>>     -- Goto (macro-hangupcall,s,9)
>>     -- Executing [s at macro-hangupcall:9] 
>> Hangup("OOH323/(null)-b7db8aa0", "") in new stack
>>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
>> 'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
>>   == Spawn extension (default, s, 2) exited non-zero on 
>> 'OOH323/(null)-b7db8aa0'
>>     -- Executing [h at default:1] Macro("OOH323/(null)-b7db8aa0", 
>> "hangupcall,") in new stack
>>     -- Executing [s at macro-hangupcall:1] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?skiprg") in new stack
>>     -- Goto (macro-hangupcall,s,4)
>>     -- Executing [s at macro-hangupcall:4] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?skipblkvm") in new stack
>>     -- Goto (macro-hangupcall,s,7)
>>     -- Executing [s at macro-hangupcall:7] 
>> GotoIf("OOH323/(null)-b7db8aa0", "1?theend") in new stack
>>     -- Goto (macro-hangupcall,s,9)
>>     -- Executing [s at macro-hangupcall:9] 
>> Hangup("OOH323/(null)-b7db8aa0", "") in new stack
>>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 
>> 'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
>>   == Spawn extension (default, h, 1) exited non-zero on 
>> 'OOH323/(null)-b7db8aa0'
>>
>> *Here is also the content of my extensions_custom.conf:*
>> [general]
>> static=yes
>> autofallthrough=yes
>>
>> [internal]
>> exten => 1000,1,Dial(SIP/1000)
>> exten => 1000,2,HangUp()
>>
>> exten => _XXXX,1,Dial(H323/${EXTEN}@Avaya)
>> exten => _XXXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya)
>> exten  => _XXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya)
>>
>> *Here is also the content of my ooh323.conf:*
>> [general]
>> faststart=yes
>> h245tunneling=yes
>> gatekeeper=DISABLE
>> bindaddr=10.1.129.231
>> port=1720
>> callerID="ALT Asterisk PBX"
>> progress_setup=8
>> progress_alert=8
>> disallow=all
>> allow=all
>> dtmfmode=inband
>> faststart=yes
>> context=internal
>>
>> [Avaya]
>> type=friend
>> context=internal
>> host=10.1.129.247
>> port=1720
>> canreinvite=no
>> disallow=all
>> allow=alaw
>> dtmfmode=inband
>>
>> *Here is also the content of sip_custom.conf:*
>> [general]
>> context=internal
>> videosupport=yes
>> allow=h261
>> allow=h263
>> allow=h263p
>> bindaddr=10.1.129.231
>> srvlookup=yes
>> conreinvitte=no
>>
>> [1000]
>> type=friend
>> secret=malvin123
>> host=dynamic
>> dtmfmode=inband
>> disallow=all
>> allow=all
>> nat=yes
>>
>>
>> Thanks & regards,
>> Malvin
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
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