[asterisk-users] Problem releasing call from a SIP extension

[Daniel Bareiro]

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Hi all!

Making some changes in extensions.conf to test the incoming calls so that
these are derived to a SIP extension, I found something that draws attention
to me: if I test calling to my PSTN line from a mobile phone, when take the
call from the SIP extension (softphone), if the mobile phone releases the call,
sofphone do it too without problems, but if I release the call from sofphone,
from the mobile phone I see that the call chronometer continues advancing as
if the mobile phone not yet releases the call. Which can be the problem?

Seeing in logs of the CLI, I observed the following thing:

- ---------------------------------------------------------------------------
alderamin*CLI>
    -- Starting simple switch on 'DAHDI/1-1'
[May 30 14:28:46] NOTICE[18535]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)...
[May 30 14:28:47] NOTICE[18535]: chan_dahdi.c:6830 ss_thread: Got event 2 (Ring/Answered)...
    -- Executing [s at incoming:1] Dial("DAHDI/1-1", "SIP/201|15|tT") in new stack
    -- Called 201
    -- SIP/201-09243ea8 is ringing
    -- Nobody picked up in 15000 ms
    -- Executing [s at incoming:2] Hangup("DAHDI/1-1", "") in new stack
  == Spawn extension (incoming, s, 2) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
    -- Starting simple switch on 'DAHDI/1-1'
[May 30 14:29:11] NOTICE[18544]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)...
[May 30 14:29:11] ERROR[18544]: callerid.c:564 callerid_feed: No start bit found in fsk data.
[May 30 14:29:11] WARNING[18544]: chan_dahdi.c:6870 ss_thread: CallerID feed failed: Success
[May 30 14:29:11] WARNING[18544]: chan_dahdi.c:6970 ss_thread: CallerID returned with error on
channel 'DAHDI/1-1'
    -- Executing [s at incoming:1] Dial("DAHDI/1-1", "SIP/201|15|tT") in new stack
    -- Called 201
    -- SIP/201-09243ea8 is ringing
    -- Nobody picked up in 15000 ms
    -- Executing [s at incoming:2] Hangup("DAHDI/1-1", "") in new stack
  == Spawn extension (incoming, s, 2) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
    -- Starting simple switch on 'DAHDI/1-1'
[May 30 14:29:36] NOTICE[18554]: chan_dahdi.c:6830 ss_thread: Got event 18 (Ring Begin)...
[May 30 14:29:36] ERROR[18554]: callerid.c:564 callerid_feed: No start bit found in fsk data.
[May 30 14:29:36] WARNING[18554]: chan_dahdi.c:6870 ss_thread: CallerID feed failed: Success
[May 30 14:29:36] WARNING[18554]: chan_dahdi.c:6970 ss_thread: CallerID returned with error on
channel 'DAHDI/1-1'
    -- Executing [s at incoming:1] Dial("DAHDI/1-1", "SIP/201|15|tT") in new stack
    -- Called 201
    -- SIP/201-09243ea8 is ringing
    -- SIP/201-09243ea8 answered DAHDI/1-1
  == Spawn extension (incoming, s, 1) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
- ---------------------------------------------------------------------------

The lines that I'm using in the configuration file are the following:


[incoming]
exten => s,1,Dial(SIP/201,15,tT)
exten => s,2,Hangup


I think that as timeout of the call is 15 seconds and the mobile phone still
continues calling, that causes that every 15 seconds it execute again a
"switch on 'DAHDI/1-1'". Can the message "exited non-zero on 'DAHDI/1-1'" have
relation with the problem?

I was testing calling from my cell phone to an analog telephone and if the
other person hangs before I do it, I see that in the my cell phone the call
even continues persisting so that if the person of the other endpoint take the
earphone again after to hang, we can continue speaking :-D

It will be some trick of the telephone companies to collect more with the
unwary subscribers? :-D

Regards,
Daniel

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