Core Dump on receiving a call from device with 'broken' G.722 codec.
Hi, I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a 'broken' configuration from a softphone configured for G.722, I get the warning on the cli: "We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. " when fs gets the invite, but then does a core dump when it tries to: <anti-action application="bridge" data="sofia/${regext_sipprofile}/${dialed_extension}%${domain_name}"/> Below are some of the traces and info output from before the core dump happens. I see this when I run gdb on the dumpfile. #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=<value optimized out>) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec, (gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so I wonder if anybody else has seen this behavior? This happens when the destination phone is also G.722 capable (policom). If I change the "frame per packet" setting in the softphone to 2 - All works OK (but the default is 1 - so cant risk allowing G.722 if it's going to core dump fs if a user make a wrong configuration) Best Regards Keith ************************************************************************ ************************************************************* 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998182@196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d] <.....> 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998182@196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182@196.99.88.77! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js -- <.. In this js I do http call to collect maximum talktime allowed ..> 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182@196.99.88.77 to XML[27879998154@e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998182@196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998182@196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_Event-Name: [REQUEST_PARAMS] variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] variable_FreeSWITCH-Hostname: [myfsbox] variable_FreeSWITCH-IPv4: [196.99.88.77] variable_FreeSWITCH-IPv6: [::1] variable_Event-Date-Local: [2009-03-16 14:37:58] variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] variable_Event-Date-Timestamp: [1237207078659653] variable_Event-Calling-File: [sofia_reg.c] variable_Event-Calling-Function: [sofia_reg_parse_auth] variable_Event-Calling-Line-Number: [1727] variable_sip_mailbox: [879998182] variable_sip_auth_username: [27879998182] variable_sip_auth_realm: [196.99.88.77] variable_mailbox: [879998182] variable_user_name: [27879998182] variable_domain_name: [196.99.88.77] variable_record_stereo: [true] variable_default_gateway: [verso] variable_default_areacode: [87] variable_transfer_fallback_extension: [operator] variable_sip-force-expires: [180] variable_toll_allow: [domestic,international] variable_accountcode: [tbaaaa] variable_user_context: [sprof1] variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] variable_effective_caller_id_number: [879998182] variable_outbound_caller_id_name: [879998150] variable_outbound_caller_id_number: [879998150] variable_sip_from_user: [27879998182] variable_sip_from_uri: [27879998182@196.99.88.77] variable_sip_from_host: [196.99.88.77] variable_sip_from_user_stripped: [27879998182] variable_sip_from_tag: [196.99.88.77] variable_sofia_profile_name: [sprof1] variable_sofia_profile_domain_name: [196.99.88.77] variable_sip_req_user: [0879998154] variable_sip_req_uri: [0879998154@196.99.88.77] variable_sip_req_host: [196.99.88.77] variable_sip_to_user: [0879998154] variable_sip_to_uri: [0879998154@196.99.88.77] variable_sip_to_host: [196.99.88.77] variable_sip_contact_user: [27879998182] variable_sip_contact_port: [22034] variable_sip_contact_uri: [27879998182@172.16.0.63:22034] variable_sip_contact_host: [172.16.0.63] variable_channel_name: [sofia/sprof1/27879998182@196.99.88.77] variable_sip_call_id: [xr125298731411533c30039109e1921f@192.168.10.1] variable_sip_user_agent: [BrokenPhone/1.4.2] variable_sip_via_host: [172.16.0.63] variable_sip_via_port: [22034] variable_sip_via_rport: [36745] variable_presence_id: [27879998182@196.99.88.77] variable_switch_r_sdp: [v=0 o=2787999818 2265 2267 IN IP4 172.16.0.63 s=Broken c=IN IP4 172.16.0.63 t=0 0 m=audio 29081 RTP/AVP 9 18 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=candidates:-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20 .1:29081 ] variable_outboundcontext: [setupprepaycall] variable_remote_media_ip: [172.16.0.63] variable_remote_media_port: [29081] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_local_media_ip: [196.99.88.77] variable_local_media_port: [16932] variable_endpoint_disposition: [EARLY MEDIA] variable_sip_nat_detected: [true] variable_api_hangup_hook: [jsapi::completecall.js] variable_talktime: [6870] variable_action: [allow] variable_status: [allowed] variable_integer: [102] variable_fraction: [85] variable_saytalktime: [60:0] variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] variable_max_forwards: [67] variable_current_application: [info] 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 switch_core_codec_init() Codec G722 Exists but not at the desired implementation. 8000hz 820ms 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 sofia_glue_tech_set_codec() Can't load codec? 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182@196.99.88.77 has no read codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 switch_core_session_write_frame() sofia/sprof1/27879998182@196.99.88.77 has no write codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182@196.99.88.77 has no read codec. 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182@196.99.88.77 to XML[8154@toregext] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->8154 in context toregext 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/27879998182.2009-03-16- 14-37-59.wav 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features <... output from info application ...> 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/voxwan/8154 [48c2f400-1227-11de-8b8e-1789e43c417d] Segmentation fault (core dumped)
Hi, You should be reporting this on JIRA ( see http://wiki.freeswitch.org/wiki/Reporting_Bugs ) Also please include a "bt", not just "frame 1" as it doesnt give out much information. Math On 16-Mar-09, at 9:36 AM, Keith Laaks wrote:
Hi,
I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a ‘broken’ configuration from a softphone configured for G. 722, I get the warning on the cli:
“We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. “ when fs gets the invite, but then does a core dump when it tries to:
<anti-action application="bridge" data="sofia/${regext_sipprofile}/$ {dialed_extension}%${domain_name}"/>
Below are some of the traces and info output from before the core dump happens.
I see this when I run gdb on the dumpfile.
#0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ 8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=<value optimized out>) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec,
(gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so
I wonder if anybody else has seen this behavior?
This happens when the destination phone is also G.722 capable (policom). If I change the “frame per packet” setting in the softphone to 2 – All works OK (but the default is 1 – so cant risk allowing G.722 if it’s going to core dump fs if a user make a wrong configuration)
Best Regards
Keith
*************************************************************************************************************************************
2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998182@196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d]
<.....>
2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998182@196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182@196.99.88.77 ! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634
2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js --
<.. In this js I do http call to collect maximum talktime allowed ..>
2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182@196.99.88.77 to XML[27879998154@e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998182@196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998182@196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_Event-Name: [REQUEST_PARAMS] variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] variable_FreeSWITCH-Hostname: [myfsbox] variable_FreeSWITCH-IPv4: [196.99.88.77] variable_FreeSWITCH-IPv6: [::1] variable_Event-Date-Local: [2009-03-16 14:37:58] variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] variable_Event-Date-Timestamp: [1237207078659653] variable_Event-Calling-File: [sofia_reg.c] variable_Event-Calling-Function: [sofia_reg_parse_auth] variable_Event-Calling-Line-Number: [1727] variable_sip_mailbox: [879998182] variable_sip_auth_username: [27879998182] variable_sip_auth_realm: [196.99.88.77] variable_mailbox: [879998182] variable_user_name: [27879998182] variable_domain_name: [196.99.88.77] variable_record_stereo: [true] variable_default_gateway: [verso] variable_default_areacode: [87] variable_transfer_fallback_extension: [operator] variable_sip-force-expires: [180] variable_toll_allow: [domestic,international] variable_accountcode: [tbaaaa] variable_user_context: [sprof1] variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] variable_effective_caller_id_number: [879998182] variable_outbound_caller_id_name: [879998150] variable_outbound_caller_id_number: [879998150] variable_sip_from_user: [27879998182] variable_sip_from_uri: [27879998182@196.99.88.77] variable_sip_from_host: [196.99.88.77] variable_sip_from_user_stripped: [27879998182] variable_sip_from_tag: [196.99.88.77] variable_sofia_profile_name: [sprof1] variable_sofia_profile_domain_name: [196.99.88.77] variable_sip_req_user: [0879998154] variable_sip_req_uri: [0879998154@196.99.88.77] variable_sip_req_host: [196.99.88.77] variable_sip_to_user: [0879998154] variable_sip_to_uri: [0879998154@196.99.88.77] variable_sip_to_host: [196.99.88.77] variable_sip_contact_user: [27879998182] variable_sip_contact_port: [22034] variable_sip_contact_uri: [27879998182@172.16.0.63:22034] variable_sip_contact_host: [172.16.0.63] variable_channel_name: [sofia/sprof1/27879998182@196.99.88.77] variable_sip_call_id: [xr125298731411533c30039109e1921f@192.168.10.1] variable_sip_user_agent: [BrokenPhone/1.4.2] variable_sip_via_host: [172.16.0.63] variable_sip_via_port: [22034] variable_sip_via_rport: [36745] variable_presence_id: [27879998182@196.99.88.77] variable_switch_r_sdp: [v=0 o=2787999818 2265 2267 IN IP4 172.16.0.63 s=Broken c=IN IP4 172.16.0.63 t=0 0 m=audio 29081 RTP/AVP 9 18 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a = candidates :-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20.1:29081 ] variable_outboundcontext: [setupprepaycall] variable_remote_media_ip: [172.16.0.63] variable_remote_media_port: [29081] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_local_media_ip: [196.99.88.77] variable_local_media_port: [16932] variable_endpoint_disposition: [EARLY MEDIA] variable_sip_nat_detected: [true] variable_api_hangup_hook: [jsapi::completecall.js] variable_talktime: [6870] variable_action: [allow] variable_status: [allowed] variable_integer: [102] variable_fraction: [85] variable_saytalktime: [60:0] variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] variable_max_forwards: [67] variable_current_application: [info]
2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 switch_core_codec_init() Codec G722 Exists but not at the desired implementation. 8000hz 820ms 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 sofia_glue_tech_set_codec() Can't load codec? 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/ sprof1/27879998182@196.99.88.77 has no read codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 switch_core_session_write_frame() sofia/sprof1/27879998182@196.99.88.77 has no write codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/ sprof1/27879998182@196.99.88.77 has no read codec. 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182@196.99.88.77 to XML[8154@toregext] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->8154 in context toregext 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ usr/local/freeswitch/recordings/27879998182.2009-03-16-14-37-59.wav 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features
<… output from info application …>
2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/voxwan/8154 [48c2f400-1227-11de-8b8e-1789e43c417d] Segmentation fault (core dumped)
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http://wiki.freeswitch.org/wiki/Reporting_Bugs Keith, Please read the link above... open a jira and collect a sip trace of this also and "attach" it. /b On Mar 16, 2009, at 8:36 AM, Keith Laaks wrote:
Hi,
I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a ‘broken’ configuration from a softphone configured for G. 722, I get the warning on the cli:
“We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. “ when fs gets the invite, but then does a core dump when it tries to:
<anti-action application="bridge" data="sofia/${regext_sipprofile}/$ {dialed_extension}%${domain_name}"/>
Below are some of the traces and info output from before the core dump happens.
I see this when I run gdb on the dumpfile.
#0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ 8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=<value optimized out>) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec,
(gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so
I wonder if anybody else has seen this behavior?
This happens when the destination phone is also G.722 capable (policom). If I change the “frame per packet” setting in the softphone to 2 – All works OK (but the default is 1 – so cant risk allowing G.722 if it’s going to core dump fs if a user make a wrong configuration)
Best Regards
Keith
participants (3)
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Brian West
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Keith Laaks
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Mathieu Rene