[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, I’ve 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

 

Here’s 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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