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 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.
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