[Asterisk-Users] Segmentation Fautl / Core Dump with G.729
Jorge Alayon
j.alayon at ses.com.ar
Wed Jun 1 09:04:19 MST 2005
Hello,
Has anyone experienced a segmentation fault in asterisk for using G729
against an AS5300 in SIP ?
I'm having this problem. Also, any other codec I use gives me incompatible
media (can be read in SIP DEBUG messages).
AS5300 configured for multiple codecs, so is Asterisk.
Tried G711u/A G723 and G.729. Any clues ?
Regards,
Jorge A.
Info:
Asterisk ver 1.0.7 stable
Using AMPortal 1.0.0.8
SIP.CONF
----------------------------------------------------------------------------
---
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
allow=g729
allow=g723
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
language=es
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
----------------------------------------------------------------------------
---
SIP_ADDITIONAL.CONF
----------------------------------------------------------------------------
---
[as5300]
type=peer
qualify=yes
host=xxx.xxx.xxx.xxx (AS5300 box)
----------------------------------------------------------------------------
---
AS5300 relevant Config
----------------------------------------------------------------------------
---
...
!
voice class codec 1010
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g723ar63
codec preference 4 g723r63
!
...
!
dial-peer voice 1010 voip
destination-pattern 85..
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1010
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx (ASterisk Box)
dtmf-relay cisco-rtp rtp-nte h245-signal h245-alphanumeric
!
----------------------------------------------------------------------------
---
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