Sip Rtp
2003-Aug-08 07:28 UTC
Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]
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@inaccessnetworks.com> To: <asterisk-users@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.Iam> > 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 theimportant> sections of extensions.conf. > > > But the same scenerio works fine when i use 723codec> > in the ATA .I can dial > > the number and extension very well/(I have 723support> > in the * ). > > But now problem comes in the outbound as when iuse a> > extension like > > exten=>12,1,Dial(OH323/12) > > Then the call goes through but i don't hear anyvoice.> > So my two problems are > > 1.Why asterisk gives seg. fault when i dial extenon> > 711 codec from ATA > > 2.Why can't i hear voice from * to ATA when i use723> > 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@lists.digium.com >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
Michael Manousos
2003-Aug-08 08:42 UTC
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@inaccessnetworks.com> > To: <asterisk-users@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@lists.digium.com >> > > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Sip Rtp
2003-Aug-11 04:36 UTC
Re2: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]
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@inaccessnetworks.com> To: <asterisk-users@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 iforgot> > 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 inRTP_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@inaccessnetworks.com> > > To: <asterisk-users@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 callthen> >>>then as soon as i dial an > >>>extension the * crashes with 'segmentationfault'.> >> > >>More information is needed. > >>You should provide a backtrace of the core file, > >>the screen log of Asterisk (generated whenexecuted> >>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 betweenthe> >>>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@lists.digium.com > >> > > > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > __________________________________ > > Do you Yahoo!? > > Yahoo! SiteBuilder - Free, easy-to-use web sitedesign software> > http://sitebuilder.yahoo.com > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >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