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

[Michael Manousos]

Sip Rtp wrote:
> 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"
> <manousos at inaccessnetworks.com>
> To: <asterisk-users at 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 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.
>>>>
>>>>
>>>>
>>>>_______________________________________________
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>>>>Asterisk-Users at lists.digium.com
>>>>
>>>
>>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>>>
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