Thanks. I want to dial-out to PSTN using Asterisk Server via Avaya Phone using Cordia VoIP Service provider. How can I achieve it using the same code below?

Regards,
Malvin

On 7/13/2011 4:59 PM, DHAVAL INDRODIYA wrote:
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 <mailto: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
    <mailto: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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New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

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