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

[Sip Rtp]

Hello Michael,

Yes i tried these values and also there is no segfault
except in case of
G711-ulaw alaw.
So there is no change in the situtaion.
Any more idea ..

Rgds
SIP RTP
----- Original Message -----
From: "Michael Manousos"
<manousos at inaccessnetworks.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, August 08, 2003 9:12 PM
Subject: Re: Re2: [Asterisk-Users]
Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]


>
> Try to set the "frames" option in section [codecs]
> to a reasonable value, say 20 for G711, 2 for G7231,
> 4 for GSM.
>
> Also, do you get segfaults when you try the same
> with just one codec enabled?
>
>
> Michael.
>
>
> Sip Rtp wrote:
> > 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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