[asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0

nik600 nik600 at gmail.com
Wed Sep 12 02:17:09 CDT 2007


hi, here is a more verbose log, obtained from enebling debug console
from logger.conf

Sep 12 12:33:01 DEBUG[3631]: chan_sip.c:10709 handle_request_invite:
Checking SIP call limits for device
Sep 12 12:33:01 DEBUG[3631]: chan_sip.c:6282 build_route: build_route:
Contact hop: <sip:172.20.0.80>
    -- Executing Answer("SIP/172.20.0.80-0819e0b8", "") in new stack
    -- Executing Dial("SIP/172.20.0.80-0819e0b8",
"SIP/caller at 172.20.0.75:5090") in new stack
Sep 12 12:33:03 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on 'D3A1DFC62C014FE9A40A4783E411AEF90xac140050' of
Response 1: Match Found
Sep 12 12:33:07 DEBUG[3648]: chan_sip.c:2085 sip_call: Outgoing Call for caller
    -- Called caller at 172.20.0.75:5090
Sep 12 12:33:07 DEBUG[3648]: chan_sip.c:3055 sip_rtp_read: Oooh,
format changed to 2
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1468 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'65a4816770971fbb516b87f4522be078 at 172.20.0.74' Request 102: Found
    -- SIP/172.20.0.75:5090-081a35f8 is ringing
Sep 12 12:33:07 DEBUG[3648]: channel.c:2105 ast_indicate: Driver for
channel 'SIP/172.20.0.80-0819e0b8' does not support indication 3,
emulating it
Sep 12 12:33:07 DEBUG[3648]: channel.c:1777 ast_settimeout: Scheduling
timer at 160 sample intervals
Sep 12 12:33:07 DEBUG[3648]: channel.c:2044 ast_read: Generator got
voice, switching to phase locked mode
Sep 12 12:33:07 DEBUG[3648]: channel.c:1777 ast_settimeout: Scheduling
timer at 0 sample intervals
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1392 __sip_ack: Acked pending invite 102
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on '65a4816770971fbb516b87f4522be078 at 172.20.0.74' of
Request 102: Match Found
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:3785 process_sdp: Oooh, we
need to change our formats since our peer supports only 0x8 (alaw) and
not 0x4 (ulaw)
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:6282 build_route: build_route:
Contact hop: <sip:172.20.0.75:5090>
    -- SIP/172.20.0.75:5090-081a35f8 answered SIP/172.20.0.80-0819e0b8
Sep 12 12:33:07 DEBUG[3648]: channel.c:1777 ast_settimeout: Scheduling
timer at 0 sample intervals
    -- Attempting native bridge of SIP/172.20.0.80-0819e0b8 and
SIP/172.20.0.75:5090-081a35f8
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1468 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'D3A1DFC62C014FE9A40A4783E411AEF90xac140050' Request 102: Found
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1392 __sip_ack: Acked pending invite 102
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on 'D3A1DFC62C014FE9A40A4783E411AEF90xac140050' of
Request 102: Match Found
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:3785 process_sdp: Oooh, we
need to change our formats since our peer supports only 0x8 (alaw) and
not 0x2 (gsm)
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:6282 build_route: build_route:
Contact hop: <sip:172.20.0.80>
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1392 __sip_ack: Acked pending invite 103
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on '65a4816770971fbb516b87f4522be078 at 172.20.0.74' of
Request 103: Match Found
    -- Started music on hold, class 'default', on channel
'SIP/172.20.0.80-0819e0b8'
Sep 12 12:33:07 DEBUG[3631]: channel.c:1777 ast_settimeout: Scheduling
timer at 160 sample intervals
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:6225 build_route: build_route:
Retaining previous route: <sip:172.20.0.75:5090>
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1468 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'D3A1DFC62C014FE9A40A4783E411AEF90xac140050' Request 103: Found
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1392 __sip_ack: Acked pending invite 103
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on 'D3A1DFC62C014FE9A40A4783E411AEF90xac140050' of
Request 103: Match Found
Sep 12 12:33:07 DEBUG[3631]: chan_sip.c:6225 build_route: build_route:
Retaining previous route: <sip:172.20.0.80>
Sep 12 12:33:07 DEBUG[3648]: channel.c:2044 ast_read: Generator got
voice, switching to phase locked mode
Sep 12 12:33:08 DEBUG[3648]: channel.c:1777 ast_settimeout: Scheduling
timer at 0 sample intervals
Sep 12 12:33:08 DEBUG[3631]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on '65a4816770971fbb516b87f4522be078 at 172.20.0.74' of
Request 103: Match Not Found
    -- Stopped music on hold on SIP/172.20.0.80-0819e0b8
    -- Started music on hold, class 'default', on channel
'SIP/172.20.0.80-0819e0b8'
Sep 12 12:33:08 DEBUG[3631]: channel.c:1777 ast_settimeout: Scheduling
timer at 160 sample intervals
Sep 12 12:33:08 DEBUG[3631]: chan_sip.c:6225 build_route: build_route:
Retaining previous route: <sip:172.20.0.75:5090>
Sep 12 12:33:08 DEBUG[3648]: channel.c:2044 ast_read: Generator got
voice, switching to phase locked mode
Sep 12 12:33:08 DEBUG[3648]: channel.c:1777 ast_settimeout: Scheduling
timer at 0 sample intervals
Sep 12 12:33:13 DEBUG[3648]: channel.c:3637 ast_channel_bridge:
Returning from native bridge, channels: SIP/172.20.0.80-0819e0b8,
SIP/172.20.0.75:5090-081a35f8
Sep 12 12:33:13 DEBUG[3648]: chan_sip.c:2450 sip_hangup:
update_call_counter(caller) - decrement call limit counter
Sep 12 12:33:13 DEBUG[3648]: app_dial.c:1661 dial_exec_full: Exiting
with DIALSTATUS=ANSWER.
  == Spawn extension (default, 1002, 2) exited non-zero on
'SIP/172.20.0.80-0819e0b8'
    -- Stopped music on hold on SIP/172.20.0.80-0819e0b8


On 9/6/07, nik600 <nik600 at gmail.com> wrote:
> yes, i've tried asterisk -rvvvvvvvvvvvvvvvv
>
> i've also tried sip debug, but i can't reach any error... only that
> the cmmunication is finished.
>
> On 9/6/07, Shonga_Kerz <shonga_kerz at yahoo.ca> wrote:
> > Have you tried asterisk -rvvv?
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of nik600
> > Sent: Wednesday, September 05, 2007 9:14 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero
> > on 'SIP/host-0819d0d0
> >
> > Hi
> >
> > i generate a call from the dialplan in this mode:
> >
> > exten => 1002,1,Answer()
> > exten => 1002,2,Dial(SIP/user at host)
> >
> > the call is generated, but after some seconds it is interrupted, here
> > the asterisk log:
> >
> > *CLI>     -- Executing Answer("SIP/host1-0819d0d0", "") in new stack
> >     -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new stack
> >     -- Called caller at host
> >     -- SIP/host-081a2610 is ringing
> >     -- SIP/host-081a2610 answered SIP/host1-0819d0d0
> >     -- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610
> >   == Spawn extension (default, 1002, 2) exited non-zero on
> > 'SIP/host-0819d0d0'
> >
> > i've enabled sip debug, but nothing interesing has been showed
> >
> > host1 is an SJphone and host is a software that implements SIP protocol.
> >
> > Can you help me to guess where is the problem?
> >
> > if i try to create a call from SJphone 2 SJphone all works fine.
> >
> > Is possible that exists a problem in asterisk ?
> > where ? how can i find it ?
> >
> > thanks to all
> >
> > --
> > /*************/
> > nik600
> > https://sourceforge.net/projects/ccmanager
> > https://sourceforge.net/projects/nikstresser
> >
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> --
> /*************/
> nik600
> https://sourceforge.net/projects/ccmanager
> https://sourceforge.net/projects/nikstresser
>


-- 
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser



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