[asterisk-users] Problem dialing out with a TDB400P

[T. Shaw]

"exten => _9.,1,Dial(Zap/1/w${EXTEN:1},20) "

Should the "w" be before the number dialied?
Should it the "w" come AFTER the number dialed?





xytek at hotmail.com
"blah..."




>From: "Dante Passalacqua" <dpz at telefonica.net.pe>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: <asterisk-users at lists.digium.com>
>Subject: [asterisk-users] Problem dialing out with a TDB400P
>Date: Thu, 3 Aug 2006 17:48:23 -0500
>
>I've posted this to the users forum, but I guess the list has more traffic, 
>so I'm reposting it here.
>-----------------
>
>Hi, some 13 years ago I use to develop IVR systems based on Dialogic cards. 
>Now after many years away from CTI, I'm taking my first steps with 
>Asterisk. As I expected many concepts remain very similar, but getting a 
>hold of Asterisk is probing to challenging.
>
>[PROBLEM] Can't dial out from a SIP phone through a POTS.
>
>Setup: Suse 10.1 + Asterisk 1.2.10 + Zaptel 1.2.7 + Digium TDM400P with 4 
>FXO (only channel 1 is connected to a POTS). Relevant debug output and 
>configuration files are at the end of the post.
>
>First of all, dialing in to the TDM400P works fine, except for hangup 
>detection which seems to be related to reverse polarity issues (I'll left 
>that for later). So I know that the card is working and the POTS are 
>working.
>
>The problem arises when a SIP phone sitting in the LAN tries to dial an 
>external phone using the TDM400P to make the outbound call. The result is 
>that Asterisk tries to dial the number on the Zap channel, thinks it's 
>already been answer and bridges the call back to the SIP phone. However the 
>call is never made.
>
>Things I've already checked: HW compatibility, IRQ sharing, putting ww in 
>the Dial string. I've also hook an standardphone to the same POTS for 
>monitoring the call progress when Asterisk attempts to dial, and actually 
>what happens is that the line never goes off-hook (if I pickup the standard 
>phone right after Asterisk says it has completed the dial, I get a 
>dialtone)
>
>I have the feeling I'm missing something quite obvious, but I've been 
>struggling with this issue for quite a while, so I won't hurt asking. So 
>any help will be appreciated. Thanks in advance.
>
>-----------------------------------------------------------------------------------
>/var/log/asterisk/debug
>==================
>Aug 3 02:31:43 DEBUG[19467] devicestate.c: Changing state for SIP/prueba1 - 
>state 2 (In use)
>Aug 3 02:31:43 DEBUG[19489] pbx.c: Launching 'Dial'
>Aug 3 02:31:43 DEBUG[19489] chan_zap.c: Using channel 1
>Aug 3 02:31:43 DEBUG[19489] channel.c: Not copying variable 
>STACK-from-sip-95551234-1.
>Aug 3 02:31:43 DEBUG[19489] channel.c: Not copying variable SIPCALLID.
>Aug 3 02:31:43 DEBUG[19489] channel.c: Not copying variable SIPUSERAGENT.
>Aug 3 02:31:43 DEBUG[19489] channel.c: Not copying variable SIPDOMAIN.
>Aug 3 02:31:43 DEBUG[19489] channel.c: Not copying variable SIPURI.
>Aug 3 02:31:43 DEBUG[19489] chan_zap.c: Dialing 'w5551234'
>Aug 3 02:31:43 DEBUG[19489] chan_zap.c: Deferring dialing...
>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel Zap/1-1 to read format 
>ulaw
>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel SIP/prueba1-081964e8 to 
>write format ulaw
>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel SIP/prueba1-081964e8 to 
>read format ulaw
>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel Zap/1-1 to write format 
>ulaw
>Aug 3 02:31:43 DEBUG[19467] devicestate.c: Changing state for Zap/1 - state 
>2 (In use)
>Aug 3 02:31:43 DEBUG[19467] devicestate.c: Changing state for Zap/1 - state 
>2 (In use)
>Aug 3 02:31:43 DEBUG[19490] app_queue.c: Device 'SIP/prueba1' changed to 
>state '2' (In use) but we don't care because they're not a member of any 
>queue.
>Aug 3 02:31:43 DEBUG[19491] app_queue.c: Device 'Zap/1' changed to state 
>'2' (In use) but we don't care because they're not a member of any queue.
>Aug 3 02:31:43 DEBUG[19492] app_queue.c: Device 'Zap/1' changed to state 
>'2' (In use) but we don't care because they're not a member of any queue.
>Aug 3 02:31:44 DEBUG[19489] chan_zap.c: Exception on 14, channel 1
>Aug 3 02:31:44 DEBUG[19489] chan_zap.c: Got event Hook Transition 
>Complete(12) on channel 1 (index 0)
>Aug 3 02:31:46 DEBUG[19489] chan_zap.c: Exception on 14, channel 1
>Aug 3 02:31:46 DEBUG[19489] chan_zap.c: Got event Dial Complete(9) on 
>channel 1 (index 0)
>Aug 3 02:31:46 DEBUG[19489] chan_zap.c: Enabled echo cancellation on 
>channel 1
>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel SIP/prueba1-081964e8 to 
>read format ulaw
>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel Zap/1-1 to write format 
>ulaw
>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel Zap/1-1 to read format 
>ulaw
>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel SIP/prueba1-081964e8 to 
>write format ulaw
>Aug 3 02:31:46 DEBUG[19489] chan_sip.c: sip_answer(SIP/prueba1-081964e8)
>
>/etc/zaptel.conf
>============
>fxsls=1
>loadzone=es
>defaultzone=es
>
>/etc/asterisk/zapata.conf
>====================
>[channels]
>echocancel=yes
>signalling=fxs_ls
>context=inbound
>busydetect=no
>callprogress=no
>faxdetect=no
>immediate=no
>channel => 1
>
>/etc/asterisk/extensions.conf
>=======================
>[from-sip]
>ignorepat => 9
>exten => 6001,1,Dial(SIP/prueba1)
>
>exten => _9.,1,Dial(Zap/1/w${EXTEN:1},20)
>exten => _9.,2,Congestion()
>exten => _9.,102,Congestion()


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