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Dante Passalacqua

2006-Aug-03 15:48 UTC

[asterisk-users] Problem dialing out with a TDB400P

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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T. Shaw

2006-Aug-03 16:37 UTC

head link

[asterisk-users] Problem dialing out with a TDB400P

"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@hotmail.com
"blah..."



>From: "Dante Passalacqua" <dpz@telefonica.net.pe>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users@lists.digium.com>
>To: <asterisk-users@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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>
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asterisk users - Aug 2006 - Problem dialing out with a TDB400P

[asterisk-users] Problem dialing out with a TDB400P

[asterisk-users] Problem dialing out with a TDB400P