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. > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
