[asterisk-bugs] [JIRA] (ASTERISK-25504) Asterisk with pjsip driver crashes codec related?

[Carl Fortin (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25504?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228046#comment-228046 ] 

Carl Fortin edited comment on ASTERISK-25504 at 11/4/15 1:20 PM:
-----------------------------------------------------------------

See Endpoint_detail.txt


was (Author: phonefxg):
Here's the detailed configuration for the endpoint that seems to be the problem:

{noformat}
ParameterName                 : ParameterValue
 ==========================================================
 100rel                        : yes
 accountcode                   : garneau
 aggregate_mwi                 : true
 allow                         : (ulaw|g722)
 allow_subscribe               : true
 allow_transfer                : true
 aors                          : 01-A-A4934CFE892B
 auth                          : 01-A-A4934CFE892B
 call_group                    :
 callerid                      : "Philippe Thibault" <2855>
 callerid_privacy              : allowed_not_screened
 callerid_tag                  :
 connected_line_method         : invite
 context                       : garneau-interne
 cos_audio                     : 6
 cos_video                     : 0
 device_state_busy_at          : 3
 direct_media                  : true
 direct_media_glare_mitigation : none
 direct_media_method           : invite
 disable_direct_media_on_nat   : true
 dtls_ca_file                  :
 dtls_ca_path                  :
 dtls_cert_file                :
 dtls_cipher                   :
 dtls_fingerprint              : SHA-256
 dtls_private_key              :
 dtls_rekey                    : 0
 dtls_setup                    : active
 dtls_verify                   : No
 dtmf_mode                     : rfc4733
 fax_detect                    : false
 force_avp                     : false
 force_rport                   : true
 from_domain                   : xxx.xxx.xxx.x
 from_user                     :
 g726_non_standard             : false
 ice_support                   : false
 identify_by                   : username
 inband_progress               : false
 language                      : fr
 mailboxes                     : 2855 at default
 media_address                 :
 media_encryption              : no
 media_encryption_optimistic   : false
 media_use_received_transport  : false
 message_context               :
 moh_suggest                   : default
 mwi_from_user                 :
 named_call_group              :
 named_pickup_group            :
 one_touch_recording           : false
 outbound_auth                 : 01-A-A4934CFE892B
 outbound_proxy                :
 pickup_group                  :
 record_off_feature            : automixmon
 record_on_feature             : automixmon
 rewrite_contact               : false
 rpid_immediate                : false
 rtp_engine                    : asterisk
 rtp_ipv6                      : false
 rtp_keepalive                 : 0
 rtp_symmetric                 : false
 rtp_timeout                   : 0
 rtp_timeout_hold              : 0
 sdp_owner                     : -
 sdp_session                   : Asterisk
 send_diversion                : true
 send_pai                      : true
 send_rpid                     : true
 set_var                       :
 srtp_tag_32                   : false
 sub_min_expiry                : 0
 t38_udptl                     : false
 t38_udptl_ec                  : none
 t38_udptl_ipv6                : false
 t38_udptl_maxdatagram         : 400
 t38_udptl_nat                 : false
 timers                        : yes
 timers_min_se                 : 90
 timers_sess_expires           : 1800
 tone_zone                     : us
 tos_audio                     : 184
 tos_video                     : 0
 transport                     : transport-udp
 trust_id_inbound              : false
 trust_id_outbound             : false
 use_avpf                      : false
 use_ptime                     : false
 user_eq_phone                 : false
{noformat}

[Edit by Rusty - formatting.. also please don't post large debug or config excerpts inside comments. Please attach them to the issue as .txt files. Thanks!]


> Asterisk with pjsip driver crashes codec related?
> -------------------------------------------------
>
>                 Key: ASTERISK-25504
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25504
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.5.0
>         Environment: Asterisk Realtime 13.5 PJSIP Driver 
> mysql Ver 14.14 
> pjproject 2.3 
> spandsp 0.0.6 
> jansson 2.7 
> CentOS 6.6 64 bits on Vmware 
> Number of endpoints : > 700 
> Numbers of calls : 3000/day 
> Our Hardware: 
> Phones : Cisco SPA514G FW: 7.5.7 
> ATA : Audiocodes MP124 
> T1 : Mediatrix 3532 ISDN to SIP gateway 
> CPU : Quadcore Intel(R) Xeon(R) CPU E5-2650 
> RAM : 3 GB
>            Reporter: Carl Fortin
>         Attachments: backtrace.txt, My_debug_log.txt
>
>
> We are running asterisk 13.5 pjsip in a production environment and I had 2 crashes yesterday.  It happened shortly after enabling the g722 codec to a couples of phones in our ps_endpoints database. I'm not sure if the problem is coming from this modification but the same phone with the g722 codec made asterisk crashed 2 times.
> Can someone look and my backtrace and my full debug log to see if it could be related to a codec negotiation?



--
This message was sent by Atlassian JIRA
(v6.2#6252)