[asterisk-dev] segmentation fault with chan_h323, asterisk , Open H.323 version v1.18.0, PWLib v1.10.0

nik600 nik600 at gmail.com
Mon Jul 21 03:48:09 CDT 2008

Hi to all

i've installed asterisk
and configured with Open H.323 version v1.18.0, PWLib v1.10.0.

I've installed them but i'm experiencing this problem:

i've configured in h323.conf 2 peers:
one to an 3.3 CCM Cisco
one to an 4.2 CCM Cisco

each CCM has the preferred codec set up as G711 ulaw.

I can forward calls from a SIP account on asterisk (using Xten-xlite
as softphone) to both the peers and talk with their extensions without
any problem.

I can forward calls from both the peers to Asterisk (and for example
place the call in queue or background some sound files)


when i try to call from the CCM 3.3 to Asterisk, and then dial from
the dialplan a SIP account, when the SIP user accept the call (using
Xten-xlite as softphone) asterisk dies with a segmentation fault

This happend only with CCM 3.3, with 4.2 there is no problem.

I've got a backtrace of the error, it seems a codec problem, as the
parameter passed to ast_rtp_new_source is null.

#0 ast_rtp_new_source (rtp=0x0) at rtp.c:2002 2002 rtp->set_marker_bit
= 1; (gdb) bt
#0 ast_rtp_new_source (rtp=0x0) at rtp.c:2002
#1 0xb6cfc346 in oh323_indicate (c=0x8205ea0, condition=20, data=0x0,
datalen=0) at chan_h323.c:919
#2 0x08081ece in ast_indicate_data (chan=0x8205ea0, condition=20,
data=0x0, datalen=0) at channel.c:2372
#3 0x0808698c in ast_channel_bridge (c0=0x8205ea0, c1=0x820acf8,
config=0xb60e0de8, fo=0xb60dff38, rc=0xb60dff34) at channel.c:2358
#4 0xb6fad295 in ast_bridge_call (chan=0x8205ea0, peer=0x820acf8,
config=0xb60e0de8) at res_features.c:1422
#5 0xb6ae0893 in dial_exec_full (chan=0x8205ea0, data=0xb6ae26fb,
peerflags=0xb60e0ea4, continue_exec=0x0) at app_dial.c:1699
#6 0xb6ae1cd2 in dial_exec (chan=0x8205ea0, data=0xb60e2f18) at app_dial.c:1753
#7 0x080c6f36 in pbx_extension_helper (c=0x8205ea0, con=0x0,
context=0x8206020 "from-h323", exten=0x8206070 "54", priority=1,
label=0x0, callerid=0x8205830 "419", action=E_SPAWN) at pbx.c:537
#8 0x080c8fb5 in __ast_pbx_run (c=0x8205ea0) at pbx.c:2317
#9 0x080c9e7e in pbx_thread (data=0x8205ea0) at pbx.c:2636
#10 0x080f8fab in dummy_start (data=0x8205ce8) at utils.c:895
#11 0xb7f56383 in start_thread () from /lib/libpthread.so.0
#12 0xb731905e in clone () from /lib/libc.so.6

Can someone help me please?


More information about the asterisk-dev mailing list