[Asterisk-Dev] SIP PA168-based phones calling-in crashes relatedtodivision by 0 at __ast_dsp_silence

[Lin Rongrong]

yes, exactly!
we will add "nattraversal" option a "never" select item in 1.39 for asterisk
situations.

Woody
Welcome to use more PA1688 resources on internet:
Mailing list in Yahoo: http://groups.yahoo.com/group/pa1688/
Discussion Forum: http://bbs.chinagk.org/dispbbs.asp?boardID=14

----- Original Message ----- 
From: "Eric Wieling" <eric at fnords.org>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Monday, October 11, 2004 7:15 AM
Subject: Re: [Asterisk-Dev] SIP PA168-based phones calling-in crashes 
relatedtodivision by 0 at __ast_dsp_silence


> Lin Rongrong wrote:
>>> From PA1688 software point of view, the 0 length error rtp packet
>>
>> was sent when rtp opens to solve NAT udp mapping problem. So maybe
>> the check of 0 length rtp packet should be done earlier in asterisk
>> software, when any rtp data packet received, check the length, and
>> if the length is 0, ignore the whole packet.
>>
>> When you are sure that your PBX and phone works on the same LAN
>> and no NAT problem at all, you can also change the PA1688 software
>> from sending this packet, do not call RtpSendNatHint() in v_task.c
>> is ok. We will also consider to add an option to prevent this from
>> called in 1.39.
>>
>
> maybe nat=never would be useful in this case
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>