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

[Michael Manousos]

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
>>
> 
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> 
> 
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