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[Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec.


 
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keithl at voxtelecom.c...
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PostPosted: Mon Mar 16, 2009 8:56 am    Post subject: [Freeswitch-users] Core Dump on receiving a call from device Reply with quote

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)
Back to top
mrene_lists at avgs.ca
Guest





PostPosted: Mon Mar 16, 2009 8:58 am    Post subject: [Freeswitch-users] Core Dump on receiving a call from device Reply with quote

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:
Quote:
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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) [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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email])
2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182@196.99.88.77 ([email]sofia/sprof1/27879998182@196.99.88.77[/email])!
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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) 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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email])]
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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email])]
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 (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 (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 (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 (27879998182@172.16.0.63):22034]
variable_sip_contact_host: [172.16.0.63]
variable_channel_name: [sofia/sprof1/27879998182@196.99.88.77 ([email]sofia/sprof1/27879998182@196.99.88.77[/email])]
variable_sip_call_id: [xr125298731411533c30039109e1921f@192.168.10.1 (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 (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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) 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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) 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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) 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 ([email]sofia/sprof1/27879998182@196.99.88.77[/email]) 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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brian at freeswitch.org
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PostPosted: Mon Mar 16, 2009 8:59 am    Post subject: [Freeswitch-users] Core Dump on receiving a call from device Reply with quote

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:
Quote:
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
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