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 ..
The problem seems to be in OpenH323. It tries to construct a bigger RTP frame, than the size it has already allocated. Check the number of frames that your H.323 terminal sends per RTP packet and put this value in the "frames" conf variable.
Michael.
Rgds SIP RTP ----- Original Message ----- From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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
http://lists.digium.com/mailman/listinfo/asterisk-userscodecs 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.
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