Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)?
Any input is highly appreciated. Regards Arun -----Original Message----- From: Mark Thompson [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault with chan_oh323 This also happened to me when I was using the same codec with both oh323 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection worked. I also tried h323 instead of oh323 which works okay but you have to use earlier versions of pwlib and openh323. Mark -----Original Message----- From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] Sent: 16 July 2003 23:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault with chan_oh323 Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=<phone number>. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
