Hi everyone,
I’m having problems with my asterisk because after a call is established it sends a new invite and the SDP changes.
I’d like to know why asterisk sends this second invite after everything is negotiated and the call is working properly with audio in both ways.
In pjsip.conf i don’t see any parameter related to this second invite.
Any idea why this happens? If you need me to give you more details please let me know
Thanks in advance
You would need to provide the configuration, as well as the full trace of the SIP packets, and state the version of Asterisk.
Thank you for your reply,
I’m using asterisk 16.7
First invite and 401 reply are for authentication.
Invite sent to asterisk with autentication
100 Trying (Reply from asterisk)
200 (Reply from asterisk)
ACK sent to asterisk
After media starts to flow and having audio in both ways asterisk sends this invite
100 trying (sent to asterisk)
200 (sent to asterisk)
ACK (Reply from asterisk)
Endpoint configuration is:
pjsip show endpoint myendpoint
Endpoint: myendpoint Unavailable 0 of 1
InAuth: Myauth
Aor: myendpoint 1
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (codec2|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|h265|mpeg4|vp8|vp9|red|t140|t38|silk|silk|silk|silk)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : myendpoint
asymmetric_rtp_codec : false
auth : myauth
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid :
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : SCAIP-VoIP_CarePhones_context
cos_audio : 0
cos_video : 0
device_state_busy_at : 1
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : No
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
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : true
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_mwi_mailbox :
language :
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : true
message_context : SCAIP_message_context
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : no
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy : sip:mydomain:5060;lr
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : false
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
suppress_q850_reason_headers : false
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 :
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : true
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
Again, thank you in advance
You’re using an old version on an unsupported branch, where I know after fixes and changes have occurred in this area. I’m not going to be able to add anything additional to this post with that version in use.
Thank you for the information.
Could you tell me where i can check the fixes that have been made to this branch?
Do you recommend to update to any particular branch or version?
Again, thanks for your help
Everything is in git[1] and each release also includes a change log. Current supported branches are documented on the docs site[2] and the latest releases are in the main downloads location[3] or the website[4].
[1] GitHub - asterisk/asterisk: The official Asterisk Project repository.
[2] Asterisk Versions - Asterisk Documentation
[3] Index of /pub/telephony/asterisk/
[4] All Asterisk Versions ⋆ Asterisk
Thank you again for your help
i’ll mark it as solved
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