Summary: | ASTERISK-27825: Sometimes when the call is intercepted, the asterisk reloads | ||
Reporter: | Nikolay (tensor) | Labels: | fax pjsip |
Date Opened: | 2018-04-24 01:36:55 | Date Closed: | 2020-01-14 11:14:05.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Applications/app_directed_pickup |
Versions: | 13.18.2 | Frequency of Occurrence | Frequent |
Related Issues: | |||
Environment: | hardware: Intel(R) Core(TM) i3-2120 CPU @ 3.30GHz, 4GB RAM, CentOS Linux release 7.4.1708 (Core), kernel 3.10.0-693.5.2.el7.x86_64, asterisk version is 13.18.2 | Attachments: | ( 0) pickup.png |
Description: | Sometimes when the call is intercepted, the asterisk reloads.Log in the attachment. | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-04-24 01:36:56.219-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Nikolay (tensor) 2018-04-24 01:38:25.123-0500 Log By: George Joseph (gjoseph) 2018-04-24 09:18:08.903-0500 Thank you for the crash report. However, we need more information to investigate the crash. Please provide: 1. A backtrace generated from a core dump using the instructions provided on the Asterisk wiki [1]. 2. Specific steps taken that lead to the crash. 3. All configuration information necesary to reproduce the crash. Thanks! [1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: Nikolay (tensor) 2018-04-25 03:54:50.860-0500 1. When restarting the asterisk in this case, the dump is not generated. 2. In order to repeat this problem you just need to make a call interception, it repeats. 3. What kind of configuration information do you want? By: George Joseph (gjoseph) 2018-04-26 12:31:07.503-0500 I've tried to reproduce the issue but can't seem to. Can you provide the full endpoint configuration for the ringing extension (looks like 5815 in ytour screen shot) and the extension attempting the pickup (5811). You can use "pjsip show endpoint 5815" and "pjsip show endpoint 5811". Also can you provide the dialplan from extensions.conf and/or extensions.lua that was invoked? Specifically, I'm looking to see if you set namedpickupgroup or any other pickup related channel variables or are you simply calling Pickup() with no arguments and relying on the "pickup_group" or "named_pickup_group" parameters on the endpoints. By: Nikolay (tensor) 2018-05-04 06:06:31.337-0500 srv-platforma1*CLI> pjsip show endpoint 5815 {noformat} Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName...........................................................> Aor: <Aor............................................> <MaxContact> Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <criteria.........................> Channel: <ChannelId......................................> <State.....> <Time.....> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================== Endpoint: 5815/5815 Not in use 0 of inf InAuth: 5815/5815 Aor: 5815 10 Contact: 5815/sip:5815@10.40.1.121:5060 d9f5a723a3 Avail 35.102 Transport: udp-transport udp 0 96 0.0.0.0:5060 ParameterName : ParameterValue ========================================================================================== 100rel : no QFBusy : managers QFNoAns : managers QFUnavail : managers accountcode : acl : aggregate_mwi : true allow : (alaw|g729) allow_overlap : true allow_subscribe : true allow_transfer : true aors : 5815 asymmetric_rtp_codec : false auth : 5815 bind_rtp_to_media_address : false call_group : callerid : "Досик Наталья Сергеевна" <5815> callerid_privacy : allowed_not_screened callerid_tag : connected_line_method : invite contact_acl : context : from-local cos_audio : 0 cos_video : 0 device_state_busy_at : 0 direct_media : false direct_media_glare_mitigation : none direct_media_method : invite disable_direct_media_on_nat : false 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 : auto fax_detect : false fax_detect_timeout : 0 force_avp : false force_rport : true from_domain : from_user : g726_non_standard : false ice_support : false identify_by : username inband_progress : true incoming_mwi_mailbox : language : ru mailboxes : media_address : media_encryption : no media_encryption_optimistic : false media_use_received_transport : false message_context : moh_suggest : default mwi_from_user : mwi_subscribe_replaces_unsolicited : false named_call_group : managers named_pickup_group : managers notify_early_inuse_ringing : false one_touch_recording : false outbound_auth : outbound_proxy : pickup_group : record_off_feature : automixmon record_on_feature : automixmon refer_blind_progress : true rewrite_contact : true rpid_immediate : false rtcp_mux : false rtp_engine : asterisk rtp_ipv6 : false rtp_keepalive : 0 rtp_symmetric : false rtp_timeout : 60 rtp_timeout_hold : 300 sdp_owner : - sdp_session : Asterisk send_diversion : true send_pai : false send_rpid : false srtp_tag_32 : false sub_min_expiry : 0 subscribe_context : t38_udptl : false t38_udptl_ec : none t38_udptl_ipv6 : false t38_udptl_maxdatagram : 0 t38_udptl_nat : false timers : yes timers_min_se : 90 timers_sess_expires : 1800 tone_zone : tos_audio : 0 tos_video : 0 transport : udp-transport trust_id_inbound : false trust_id_outbound : false use_avpf : false use_ptime : false user_eq_phone : false voicemail_extension : {noformat} srv-platforma1*CLI> pjsip show endpoint 5811 {noformat} Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName...........................................................> Aor: <Aor............................................> <MaxContact> Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <criteria.........................> Channel: <ChannelId......................................> <State.....> <Time.....> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================== Endpoint: 5811/5811 In use 34 of inf InAuth: 5811/5811 Aor: 5811 10 Contact: 5811/sip:5811@10.40.1.51:5060 3df635780e Avail 31.335 Transport: udp-transport udp 0 96 0.0.0.0:5060 Channel: PJSIP/5811-00002ac7/AppQueue Up 00:00:13 Exten: 903576 CLCID: "kaluga" <084842724565> ParameterName : ParameterValue ============================================================================================ 100rel : no accountcode : acl : aggregate_mwi : true allow : (g729) allow_overlap : true allow_subscribe : true allow_transfer : true aors : 5811 asymmetric_rtp_codec : false auth : 5811 bind_rtp_to_media_address : false call_group : callerid : "Марченко Анна Николаевна" <5811> callerid_privacy : allowed_not_screened callerid_tag : connected_line_method : invite contact_acl : context : from-local cos_audio : 0 cos_video : 0 device_state_busy_at : 0 direct_media : false direct_media_glare_mitigation : none direct_media_method : invite disable_direct_media_on_nat : false 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 : auto fax_detect : false fax_detect_timeout : 0 force_avp : false force_rport : true from_domain : from_user : g726_non_standard : false ice_support : false identify_by : username inband_progress : true incoming_mwi_mailbox : language : ru mailboxes : media_address : media_encryption : no media_encryption_optimistic : false media_use_received_transport : false message_context : moh_suggest : default mwi_from_user : mwi_subscribe_replaces_unsolicited : false named_call_group : managers named_pickup_group : managers notify_early_inuse_ringing : false one_touch_recording : false outbound_auth : outbound_proxy : pickup_group : record_off_feature : automixmon record_on_feature : automixmon refer_blind_progress : true rewrite_contact : true rpid_immediate : false rtcp_mux : false rtp_engine : asterisk rtp_ipv6 : false rtp_keepalive : 0 rtp_symmetric : false rtp_timeout : 60 rtp_timeout_hold : 300 sdp_owner : - sdp_session : Asterisk send_diversion : true send_pai : false send_rpid : false set_var : srtp_tag_32 : false sub_min_expiry : 0 subscribe_context : t38_udptl : false t38_udptl_ec : none t38_udptl_ipv6 : false t38_udptl_maxdatagram : 0 t38_udptl_nat : false timers : yes timers_min_se : 90 timers_sess_expires : 1800 tone_zone : tos_audio : 0 tos_video : 0 transport : udp-transport trust_id_inbound : false trust_id_outbound : false use_avpf : false use_ptime : false user_eq_phone : false voicemail_extension : {noformat} pickup from extensions.lua {noformat} --[[ Функция для перехвата звонка в рамках pickupgroup Входные параметры - Выходные значения - ]] app.Set("OutgoingContext=to-features") channel["DB(PickUP/".. channel["CHANNEL"]:get() ..")"]:set(channel["CALLERID(num)"]:get()) app.NoOp(string.format("DB(PickUP/%s = %s", channel["CHANNEL"]:get(), channel["CALLERID(num)"]:get())) app.Set('pick_up_channel=' .. channel["UNIQUEID"]:get()) app.PickUP() end {noformat} By: George Joseph (gjoseph) 2018-05-07 07:38:01.877-0500 If you're picking up by channel you need to call PickupChan(channel_id) By: Asterisk Team (asteriskteam) 2018-05-21 12:00:02.773-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Nikolay (tensor) 2018-05-30 08:39:52.662-0500 I need to pick up calls In a named call group By: Asterisk Team (asteriskteam) 2018-05-30 08:39:53.192-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Richard Mudgett (rmudgett) 2018-06-01 16:25:37.193-0500 Thank you for the crash report. However, we need more information to investigate the crash. Please provide: 1. A backtrace generated from a core dump using the instructions provided on the Asterisk wiki [1]. 2. Specific steps taken that lead to the crash. 3. All configuration information necesary to reproduce the crash. Thanks! [1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace We really need that backtrace to go forward with this as the crash may have something to do with LUA and not pickup. Asterisk must be started with the -g option present for it to generate core files on a crash. By: Asterisk Team (asteriskteam) 2018-06-16 12:00:01.216-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |