Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]

[Sip Rtp]

Hello Michael,

Here is the  BackTrace of the program which i forgot
to attach

BACKTRACE OF Asterisk -vvc

#0  0x42074d60 in _int_realloc () from
/lib/tls/libc.so.6
#1  0x420738c4 in realloc () from /lib/tls/libc.so.6
#2  0x47c7da89 in PAbstractArray::SetSize(int) () from
/home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5
#3  0x47c7cf4d in PContainer::SetMinSize(int) () from
/home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5
#4  0x47784af3 in RTP_DataFrame::SetPayloadSize(int)
() from
/home/sip/openh323/lib/libh323_linux_x86_r.so.1.12
#5  0x4776ea76 in H323_RTPChannel::Transmit() () from
/home/sip/openh323/lib/libh323_linux_x86_r.so.1.12
#6  0x4776ba84 in H323LogicalChannelThread::Main() ()
from
/home/sip/openh323/lib/libh323_linux_x86_r.so.1.12
#7  0x47c756f1 in PThread::PX_ThreadStart(void*) ()
from
/home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5
#8  0x4002e332 in start_thread () from
/lib/tls/libpthread.so.0

Rgds
Sip Rtp




----- Original Message -----
From: "Michael Manousos"
<manousos at inaccessnetworks.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, August 08, 2003 3:56 PM
Subject: Re: [Asterisk-Users] Problem
-ATA-711-723-Oh323-Asterisk


>
> Sip Rtp wrote:
> > Hi List,
> >
> > I am facing the reverse problem as stated here.I
am
> > using ATA 186 to make
> > and recieve call to * through OH323 driver.
> > When I use G711 codec in the ATA to make call then
> > then as soon as i dial an
> > extension the * crashes with 'segmentation fault'.
>
> More information is needed.
> You should provide a backtrace of the core file,
> the screen log of Asterisk (generated when executed
> with "asterisk -vvvcdg"), your oh323.conf and the
important
> sections of extensions.conf.
>
> > But the same scenerio works fine when i use 723
codec
> > in the ATA .I can dial
> > the number and extension very well/(I have 723
support
> > in the * ).
> > But now problem comes in the outbound as when i
use a
> > extension like
> > exten=>12,1,Dial(OH323/12)
> > Then the call goes through but i don't hear any
voice.
> > So my two problems are
> > 1.Why asterisk gives seg. fault when i dial exten
on
> > 711 codec from ATA
> > 2.Why can't i hear voice from * to ATA when i use
723
> > in ATA.
> > for 2nd i think that there is mismatch between the
> > codecs  so can we change
> > the priority order of the codecs used in the * or
> > Oh323 and if yes, then
> > how?
> >
> > Please ask if any further Input is required.
> >
> > Rgds
> > Manoj K Gupta
> >
>
>
> Michael.
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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