[asterisk-users] Asterisk with Avaya

Carlos Chavez cursor at telecomabmex.com
Fri Jan 30 11:54:39 CST 2009


On Fri, 2009-01-30 at 17:17 +0000, Edwin Quijada wrote:
> Hi !
> I am trying to connect Asterisk with Avaya Definity.
> I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
> The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything
> Example
> Asterisk ---> Avaya
>  -- Executing [73133 at internal:1] Dial("SIP/59000-08203708", "H323/73133 at Avaya") in new stack
>     -- Requested transfer capability: 0x00 - SPEECH
>     -- Called 73133 at Avaya
>     -- H323/Avaya-1 is making progress passing it to SIP/59000-08203708
>     -- H323/Avaya-1 is ringing
>     -- H323/Avaya-1 answered SIP/59000-08203708
>   == Spawn extension (internal, 73133, 1) exited non-zero on 'SIP/59000-08203708'
> 
> Everything good but I cant hear anything.
> 
> 
> Avaya ----> Asterisk
>     -- Executing [59000 at internal:1] Answer("H323/ip$10.200.1.47:23924/18397", "") in new stack
>     -- Executing [59000 at internal:2] Playback("H323/ip$10.200.1.47:23924/18397", "vm-intro") in new stack
>     --  Playing 'vm-intro' (language 'en')
>     -- Executing [59000 at internal:3] Playback("H323/ip$10.200.1.47:23924/18397", "vm-goodbye") in new stack
>     --  Playing 'vm-goodbye' (language 'en')
>     -- Executing [59000 at internal:4] Playback("H323/ip$10.200.1.47:23924/18397", "vm-intro") in new stack
>     --  Playing 'vm-intro' (language 'en')
>     -- Executing [59000 at internal:5] Wait("H323/ip$10.200.1.47:23924/18397", "2") in new stack
>     -- Executing [59000 at internal:6] Hangup("H323/ip$10.200.1.47:23924/18397", "") in new stack
>   == Spawn extension (internal, 59000, 6) exited non-zero on 'H323/ip$10.200.1.47:23924/18397'
> 
> In this case just play a message but I cant hear anything again.
> This is my conf files
> 
> ==============EXTENSION===============================
> [general]
> language=en
> static=yes
> autofallthrough=yes
> 
> 
> [internal]
> ;My extension 59xxx
> ;exten => 59000,1,Dial(SIP/59000)
> ;exten => 59000,2,VoiceMail(59000 at 118218)
> ;exten => 59000,3,PlayBack(vm-goodbye)
> ;exten => 59000,4,Wait(2)
> ;exten => 59000,5,HangUp()
> 
> exten => 59000,1,Answer
> exten => 59000,2,PlayBack(vm-intro)
> exten => 59000,3,PlayBack(vm-goodbye)
> exten => 59000,4,PlayBack(vm-intro)
> exten => 59000,5,Wait(2)
> exten => 59000,6,HangUp()
> 
> exten => _7XXXX,1,Dial(H323/${EXTEN}@Avaya); Avaya Extension
> exten => _7XXXXX,1,Dial(H323/${EXTEN}@Avaya); Avaya Extension
> exten => _5XXXX,1,Dial(H323/${EXTEN}@Avaya); to call on SIP Extension
> exten => _4XXXX,1,Dial(H323/${EXTEN}@Avaya); Your extension on Avaya
> exten => _006XXXXXXXX,1,Dial(H323/${EXTEN}@Avaya); to call on mobile
> exten => _00XXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya); to call on National
> 
> 
> =======================H323==============================
> [general]
> port = 1720
> bindaddr = 0.0.0.0      ; this SHALL contain a single, valid IP address for this machine
> amaflags = AVAYA
> progress_setup = 8
> progress_alert = 8
> faststart=yes
> h245tunneling=yes
> gatekeeper = DISABLE
> 
> ;We need to conserve the main parameters to allow the h323 to call to the SIP phone
> disallow=all
> allow=ulaw
> allow=alaw
> dtmfmode=inband
> context=internal ; name of your context
> 
> 
> [Avaya]
> type=friend
> context=internal
> host=10.200.1.47   ; IP Address of your CLAN
> port=1720; port used to connect on CLAN it could be some others port regarding your configuration in signalling gr$
> disallow=all
> allow=ulaw ;alaw
> allow=alaw
> canreinvite=no
> dtmfmode=inband
> 
> 
> ========================SIP==============================
> 
> [general]
> ;context=default
> context=internal
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> videosupport=yes ; if you want activate video support
> canreinvite=no
> 
> [59000]
> type=friend
> secret=1234 ;your password
> host=dynamic
> dtmfmode=inband
> disallow=all
> allow=ulaw 
> allow=alaw
> allow=h263 ; to use a video codec if needed
> callerid="Cyril CONSTANTIN" 
> nat=yes
> 
> *-------------------------------------------------------*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-809-849-8087
> 
> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun"
> *-------------------------------------------------------*
> 
> 

	Recently I had the same problem using H323 with Cisco and I solved it
by changing "bindaddr = 0.0.0.0" to the IP address of the Asterisk
server.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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