you can edit dial-plan by adding following lines to your code
[internal]
exten => s,1,Dial(SIP/1000)
exten => s,2,HangUp()
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)
On Wed, Jul 13, 2011 at 1:35 PM, Malvin Rito
<mr...@mail.altcladding.com.ph>wrote:
> **
> 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 <mr...@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@default:1] Playback("OOH323/(null)-b7db8aa0",
> "vm-goodbye") in new stack
> -- <OOH323/(null)-b7db8aa0> Playing 'vm-goodbye.ulaw' (language 'en')
> -- Executing [s@default:2] Macro("OOH323/(null)-b7db8aa0",
> "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("OOH323/(null)-b7db8aa0",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("OOH323/(null)-b7db8aa0",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("OOH323/(null)-b7db8aa0",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s@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@default:1] Macro("OOH323/(null)-b7db8aa0",
> "hangupcall,") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("OOH323/(null)-b7db8aa0",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("OOH323/(null)-b7db8aa0",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("OOH323/(null)-b7db8aa0",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s@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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>
> --
> _____________________________________________________________________
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_____________________________________________________________________
-- 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
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