[asterisk-bugs] [JIRA] (ASTERISK-24845) pjsip send notify not working with Cisco phone
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Mon Mar 9 11:07:35 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24845?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225296#comment-225296 ]
Matt Jordan commented on ASTERISK-24845:
----------------------------------------
That seems to imply that the authentication credentials you provided were not what the server wanted.
This looks like a configuration issue.
> pjsip send notify not working with Cisco phone
> ----------------------------------------------
>
> Key: ASTERISK-24845
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24845
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 13.2.0
> Environment: CentOS release 6.6 (Final)
> Asterisk 13.2 realtime PJsip driver
> Phones: Cisco SPA514G firmware 7.5.6a
> Reporter: Carl Fortin
>
> pjsip send notify will not work on cisco phone. I have tested the chan_sip driver and it is working great. I even tested with a yealink phone with the PJsip driver and it is working.
> Here is my post in the forum:
> http://forums.asterisk.org/viewtopic.php?f=1&t=92633&sid=6525c642fac6604b2e52af33edafb676
> Here is phone's settings:
> ParameterName : ParameterValue
> ===============================================
> 100rel : yes
> accountcode :
> aggregate_mwi : true
> allow : (ulaw)
> allow_subscribe : true
> allow_transfer : true
> aors : U_9172
> auth : U_9172
> call_group :
> callerid : "Hola" <2705>
> callerid_privacy : allowed
> callerid_tag : Carl
> connected_line_method : invite
> context : garneau-interne
> cos_audio : 6
> cos_video : 0
> device_state_busy_at : 0
> direct_media : true
> 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 : info
> fax_detect : false
> force_avp : false
> force_rport : true
> from_domain :
> from_user : U_9172
> ice_support : false
> identify_by : username
> inband_progress : true
> language : fr
> mailboxes :
> media_address :
> media_encryption : none
> 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 :
> outbound_proxy :
> pickup_group :
> record_off_feature : automixmon
> record_on_feature : automixmon
> rewrite_contact : false
> rtp_engine : asterisk
> rtp_ipv6 : false
> rtp_symmetric : false
> 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 : 0
> t38_udptl_nat : false
> timers : yes
> timers_min_se : 90
> timers_sess_expires : 1800
> tone_zone :
> tos_audio : 184
> tos_video : 0
> transport : transport-udp
> trust_id_inbound : true
> trust_id_outbound : true
> use_avpf : false
> use_ptime : false
> user_eq_phone : false
> I can provide more info if needed.
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