[Asterisk-Users] incoming call release after 1 ring

[leonimar cape]

Hi, 

I have already determine the reason why my incoming
got release after one ring. The telco that I am
connected is waiting for an immediate answer
supervision from my side. Is there anyway immediate
answer supervision be included on the ISDN messages.

Thanks 



--- leonimar cape <leo_mac_ph at yahoo.com> wrote:

> Hello,
> 
> Can somebody please assist me with my problem.
> Currently I am using a Asterisk at HOme version 2.4
> with
> a TE406P digium card. One the E1 is connected to a
> telco switch via an ISDN. May issue is that may
> incoming calls in the zap channels gets disconnected
> or release after 1 ring. I really dont know what
> setting should I change to increase the timeout of
> the
> ring. I have even tried upgrading the libraries for
> the zaptel and libpri but still no avail. Any
> suggestion will greatly be appreciated. 
> 
> Thanks in advance.
> 
> Please the trace below:
> Feb  8 21:20:18 VERBOSE[6600] logger.c:     -- AGI
> Script dialparties.agi completed, returning 0
> Feb  8 21:20:18 VERBOSE[6600] logger.c:     --
> Executing Dial("Zap/105-1", "SIP/1234|60|tr") in new
> stack
> Feb  8 21:20:18 DEBUG[6600] chan_sip.c: Setting NAT
> on
> RTP to 0
> Feb  8 21:20:18 DEBUG[6600] chan_sip.c: Outgoing
> Call
> for 1234
> Feb  8 21:20:18 VERBOSE[6600] logger.c:     --
> Called
> 1234
> Feb  8 21:20:18 DEBUG[6600] chan_zap.c: Requested
> indication 3 on channel Zap/105-1
> Feb  8 21:20:18 DEBUG[4160] channel.c: Avoiding
> initial deadlock for 'Zap/105-1'
> Feb  8 21:20:18 DEBUG[4175] chan_sip.c:
> (Provisional)
> Stopping retransmission (but retaining packet) on
> '1187ae7d6be6e0761d99cf5245a9058d at 202.58.255.137'
> Request 102: Found
> Feb  8 21:20:18 DEBUG[4175] chan_sip.c:
> (Provisional)
> Stopping retransmission (but retaining packet) on
> '1187ae7d6be6e0761d99cf5245a9058d at 202.58.255.137'
> Request 102: Found
> Feb  8 21:20:18 DEBUG[4160] channel.c: Avoiding
> initial deadlock for 'SIP/1234-3a53'
> Feb  8 21:20:18 VERBOSE[6600] logger.c:     --
> SIP/1234-3a53 is ringing
> Feb  8 21:20:19 VERBOSE[4179] logger.c:     --
> Channel
> 0/12, span 4 got hangup request
> Feb  8 21:20:19 DEBUG[6600] chan_sip.c:
> update_call_counter(1234) - decrement call limit
> counter
> Feb  8 21:20:19 DEBUG[6600] chan_sip.c: Acked
> pending
> invite 102
> Feb  8 21:20:19 DEBUG[6600] chan_sip.c: Stopping
> retransmission on
> '1187ae7d6be6e0761d99cf5245a9058d at 202.58.255.137' of
> Request 102: Match Found
> Feb  8 21:20:19 DEBUG[6600] chan_sip.c: Stopping
> retransmission on
> '1187ae7d6be6e0761d99cf5245a9058d at 202.58.255.137' of
> Request 102: Match Found
> Feb  8 21:20:19 DEBUG[6600] app_dial.c: Exiting with
> DIALSTATUS=CANCEL.
> 
> Cheers!
> 
> Mac
> 
> 
> 
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