[Asterisk-Users] Mi extensions keeps ringing
    asterisk at mazone.info 
    asterisk at mazone.info
       
    Thu Feb  3 12:06:48 MST 2005
    
    
  
Hi asterisk users, I have a inssue with incoming calls with wcfxo card,
while receiving a call, Ive configured my dialplan to forward the call to
all mi home voip extensions and that works just fine, but while in the call,
after a few seconds, the pbx starts the simple switch once more and keeps
ringing the voip extensions
 log as follows:
 
----------------------------------------------------------------------------
----------------------------------------------------------------------------
-
 
    -- Starting simple switch on 'Zap/1-1'
Feb  3 12:11:17 NOTICE[6424]: chan_zap.c:5363 ss_thread: Got event 2
(Ring/Answered)...
    -- Executing Dial("Zap/1-1",
"SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack
Feb  3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create
channel of type 'IAX2'
    -- Called 2101
Feb  3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
    -- Call accepted by 192.168.200.100 (format gsm)
    -- Format for call is gsm
    -- IAX2/2101/1 is ringing
    -- IAX2/2101/1 answered Zap/1-1
Feb  3 12:11:31 NOTICE[2025]: chan_iax2.c:1371 iax2_destroy: Avoiding IAX
destroy deadlock
    -- Hungup 'IAX2/2101/1'
  == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Starting simple switch on 'Zap/1-1'
Feb  3 12:11:54 WARNING[6428]: chan_zap.c:5434 ss_thread: CallerID returned
with error on channel 'Zap/1-1'
    -- Executing Dial("Zap/1-1",
"SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack
Feb  3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create
channel of type 'IAX2'
    -- Call accepted by 192.168.200.100 (format gsm)
    -- Format for call is gsm
    -- Called 2101
Feb  3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
Feb  3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
    -- IAX2/2101/3 is ringing
    -- Hungup 'IAX2/2101/3'
  == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1'
Hungup 'Zap/1-1'
 
----------------------------------------------------------------------------
----------------------------------------------------------------------------
-
 
Also if someone tooks the physical PSTN phone line and make an outgoing call
I receive exactly the same behavior
 
Heres what I have in my Zapata.conf and extensions.conf, please any advice
to correct/handle this problem will be awesome!
 
Thanks in advance!
 
;-------------------- zapata.conf ----------------------
 
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
rxgain=10.5
txgain=-1.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling=fxs_ks
context=default
 
;-------------------- extensions.conf ----------------------
 
[general]
static=yes
writeprotect=no
 
[globals]
CONSOLE=Console/dsp
 
TRUNK=Zap/1
PACO=IAX2/2101&SIP/2102&SIP/2103
CASA=SIP/2001&SIP/2002&IAX2/2003&${PACO}
 
;----------------------- COMMON SERVICES -------------------------
 
[default] ; context used for general services
exten => s,1,Dial(${CASA},10)
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