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<DIV><SPAN class=postbody><FONT size=2>I've posted this to the users forum, but
I guess the list has more traffic, so I'm reposting it
here.</FONT></SPAN></DIV>
<DIV><SPAN class=postbody><FONT size=2>-----------------</FONT></SPAN></DIV>
<DIV><SPAN class=postbody></SPAN> </DIV>
<DIV><SPAN class=postbody><FONT size=2>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.
<BR><BR>[PROBLEM] Can't dial out from a SIP phone through a POTS. <BR><BR>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. <BR><BR>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. <BR><BR>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. <BR><BR>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) <BR><BR>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.
<BR><BR>-----------------------------------------------------------------------------------
<BR>/var/log/asterisk/debug <BR>================== <BR>Aug 3 02:31:43
DEBUG[19467] devicestate.c: Changing state for SIP/prueba1 - state 2 (In use)
<BR>Aug 3 02:31:43 DEBUG[19489] pbx.c: Launching 'Dial' <BR>Aug 3 02:31:43
DEBUG[19489] chan_zap.c: Using channel 1 <BR>Aug 3 02:31:43 DEBUG[19489]
channel.c: Not copying variable STACK-from-sip-95551234-1. <BR>Aug 3 02:31:43
DEBUG[19489] channel.c: Not copying variable SIPCALLID. <BR>Aug 3 02:31:43
DEBUG[19489] channel.c: Not copying variable SIPUSERAGENT. <BR>Aug 3 02:31:43
DEBUG[19489] channel.c: Not copying variable SIPDOMAIN. <BR>Aug 3 02:31:43
DEBUG[19489] channel.c: Not copying variable SIPURI. <BR></FONT><FONT
size=2><SPAN style="FONT-WEIGHT: bold">Aug 3 02:31:43 DEBUG[19489] chan_zap.c:
Dialing 'w5551234' <BR>Aug 3 02:31:43 DEBUG[19489] chan_zap.c: Deferring
dialing...</SPAN> <BR>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel Zap/1-1
to read format ulaw <BR>Aug 3 02:31:43 DEBUG[19489] channel.c: Set channel
SIP/prueba1-081964e8 to write format ulaw <BR>Aug 3 02:31:43 DEBUG[19489]
channel.c: Set channel SIP/prueba1-081964e8 to read format ulaw <BR>Aug 3
02:31:43 DEBUG[19489] channel.c: Set channel Zap/1-1 to write format ulaw
<BR>Aug 3 02:31:43 DEBUG[19467] devicestate.c: Changing state for Zap/1 - state
2 (In use) <BR>Aug 3 02:31:43 DEBUG[19467] devicestate.c: Changing state for
Zap/1 - state 2 (In use) <BR>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. <BR>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. <BR>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. <BR>Aug 3 02:31:44 DEBUG[19489] chan_zap.c: Exception on 14, channel
1 <BR>Aug 3 02:31:44 DEBUG[19489] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index 0) <BR>Aug 3 02:31:46 DEBUG[19489] chan_zap.c:
Exception on 14, channel 1 <BR>Aug 3 02:31:46 DEBUG[19489] chan_zap.c: Got event
Dial Complete(9) on channel 1 (index 0) <BR>Aug 3 02:31:46 DEBUG[19489]
chan_zap.c: Enabled echo cancellation on channel 1 <BR>Aug 3 02:31:46
DEBUG[19489] channel.c: Set channel SIP/prueba1-081964e8 to read format ulaw
<BR>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel Zap/1-1 to write format
ulaw <BR>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel Zap/1-1 to read
format ulaw <BR>Aug 3 02:31:46 DEBUG[19489] channel.c: Set channel
SIP/prueba1-081964e8 to write format ulaw <BR>Aug 3 02:31:46 DEBUG[19489]
chan_sip.c: sip_answer(SIP/prueba1-081964e8) <BR><BR>/etc/zaptel.conf
<BR>============ <BR>fxsls=1 <BR>loadzone=es <BR>defaultzone=es
<BR><BR>/etc/asterisk/zapata.conf <BR>==================== <BR>[channels]
<BR>echocancel=yes <BR>signalling=fxs_ls <BR>context=inbound <BR>busydetect=no
<BR>callprogress=no <BR>faxdetect=no <BR>immediate=no <BR>channel => 1
<BR><BR>/etc/asterisk/extensions.conf <BR>======================= <BR>[from-sip]
<BR>ignorepat => 9 <BR>exten => 6001,1,Dial(SIP/prueba1) <BR><BR>exten
=> _9.,1,Dial(Zap/1/w${EXTEN:1},20) <BR>exten => _9.,2,Congestion()
<BR>exten => _9.,102,Congestion()</FONT></SPAN></DIV></BODY></HTML>