Hi guys! I have a problem trying to receive calls in a Cisco 6921. Previously we have Asterisk 1.8 with chan_sip and all was fine. We upgrade to asterisk 16 with pjsip and when we call to the cisco the log shows ‘everyone is busy/congested at this time (1:0/0/1)’ and the call end. I have made captures and I see 404 not found, so I thought about NAT problems and configured the phone following Migrating from chan_sip to res_pjsip - Asterisk Project - Asterisk Project Wiki because in chan_sip it had the setting nat=yes but it didn’t work. I also tried following this guide Configuring res_pjsip to work through NAT - Asterisk Project - Asterisk Project Wiki but I have the same problem.
This is the config for the endpoint:
[
{
“id”: “503_501”,
“transport”: “transport-udp”,
“aors”: “503_501”,
“auth”: “503_501”,
“context”: “503”,
“disallow”: null,
“allow”: “ulaw;alaw”,
“direct_media”: “no”,
“connected_line_method”: null,
“direct_media_method”: null,
“direct_media_glare_mitigation”: null,
“disable_direct_media_on_nat”: null,
“dtmf_mode”: “auto”,
“external_media_address”: null,
“force_rport”: null,
“ice_support”: null,
“identify_by”: null,
“mailboxes”: “”,
“moh_suggest”: null,
“outbound_auth”: null,
“outbound_proxy”: null,
“rewrite_contact”: “yes”,
“rtp_ipv6”: null,
“rtp_symmetric”: “yes”,
“send_diversion”: null,
“send_pai”: null,
“send_rpid”: null,
“timers_min_se”: null,
“timers”: null,
“timers_sess_expires”: null,
“callerid”: null,
“callerid_privacy”: null,
“callerid_tag”: null,
“100rel”: null,
“aggregate_mwi”: null,
“trust_id_inbound”: null,
“trust_id_outbound”: null,
“use_ptime”: null,
“use_avpf”: null,
“media_encryption”: null,
“inband_progress”: null,
“call_group”: null,
“pickup_group”: null,
“named_call_group”: null,
“named_pickup_group”: null,
“device_state_busy_at”: null,
“fax_detect”: null,
“t38_udptl”: null,
“t38_udptl_ec”: null,
“t38_udptl_maxdatagram”: null,
“t38_udptl_nat”: null,
“t38_udptl_ipv6”: null,
“tone_zone”: null,
“language”: “pt_BR”,
“one_touch_recording”: null,
“record_on_feature”: null,
“record_off_feature”: null,
“rtp_engine”: null,
“allow_transfer”: “yes”,
“allow_subscribe”: “no”,
“sdp_owner”: null,
“sdp_session”: null,
“tos_audio”: null,
“tos_video”: null,
“sub_min_expiry”: null,
“from_domain”: null,
“from_user”: null,
“mwi_from_user”: null,
“dtls_verify”: null,
“dtls_rekey”: null,
“dtls_cert_file”: null,
“dtls_private_key”: null,
“dtls_cipher”: null,
“dtls_ca_file”: null,
“dtls_ca_path”: null,
“dtls_setup”: null,
“srtp_tag_32”: null,
“media_address”: null,
“redirect_method”: null,
“set_var”: null,
“cos_audio”: null,
“cos_video”: null,
“message_context”: null,
“force_avp”: null,
“media_use_received_transport”: null,
“accountcode”: null,
“user_eq_phone”: null,
“moh_passthrough”: null,
“media_encryption_optimistic”: null,
“rpid_immediate”: null,
“g726_non_standard”: null,
“rtp_keepalive”: null,
“rtp_timeout”: null,
“rtp_timeout_hold”: null,
“bind_rtp_to_media_address”: null,
“voicemail_extension”: null,
“mwi_subscribe_replaces_unsolicited”: null,
“deny”: null,
“permit”: null,
“acl”: null,
“contact_deny”: null,
“contact_permit”: null,
“contact_acl”: null,
“subscribe_context”: null,
“fax_detect_timeout”: null,
“contact_user”: null,
“preferred_codec_only”: null,
“asymmetric_rtp_codec”: null,
“rtcp_mux”: null,
“allow_overlap”: null,
“refer_blind_progress”: null,
“notify_early_inuse_ringing”: null,
“max_audio_streams”: null,
“max_video_streams”: null,
“webrtc”: null,
“dtls_fingerprint”: null,
“incoming_mwi_mailbox”: null,
“bundle”: null,
“dtls_auto_generate_cert”: null,
“follow_early_media_fork”: null,
“accept_multiple_sdp_answers”: null,
“suppress_q850_reason_headers”: null,
“trust_connected_line”: null,
“send_connected_line”: null,
“ignore_183_without_sdp”: null,
“send_history_info”: null,
“stir_shaken”: null
}
]
Asterisk is in a public server, and the Cisco is behind a NAT
Thanks!