[asterisk-bugs] [JIRA] (ASTERISK-30439) DTMF with direct media
Vitor Gomes Faria (JIRA)
noreply at issues.asterisk.org
Fri Feb 24 09:18:03 CST 2023
[ https://issues.asterisk.org/jira/browse/ASTERISK-30439?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Vitor Gomes Faria updated ASTERISK-30439:
-----------------------------------------
Description:
Hi guys,
I'm having some difficulties establishing a topology using direct media.
The issue is when im using the direct_media = true the DTMF does not work.
My topology is:
SBC with PSTN
Server 1 with Asterisk 18.2.2
Server 2 with Asterisk 18.2.2
Server 1 has an pjsip endpoint to SBC and another endpoint to a Server 2.
Server 2 has an pjsip endpoint to Server 1
Basically the call ingress by SBC endpoint in Server 1 and forward to the Server 2.
When direct_media = false on all endpoints de rtp flow between three IPs and de DTMF works fine, but when i configure the direct media = true the rtp flow just between Server 2 and SBC (was expected and is what i want) but the DTMF doesent works.
The DTMF events does not apparece in a log of any server.
The DTMF type to all endpoints is rfc4733.
Below is the configuration of an endpoint:
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (alaw|ulaw|vp8)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : xxxxxxxxx
asymmetric_rtp_codec : false
auth : xxxxxxxxx
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : unknown
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
connected_line_method : invite
contact_acl :
context : from-internal
cos_audio : 0
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_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 : xxxxxxxxx
g726_non_standard : false
ice_support : false
identify_by : username
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : pt_BR
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
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 : xxxxxxxxx
outbound_proxy :
outgoing_call_offer_pref : remote_merge
pickup_group :
preferred_codec_only : false
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 : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : 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 : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
Did I forget some setting for this to work?
was:
Hi guys,
I'm having some difficulties establishing a topology using direct media.
The issue is when im using the direct_media = true the DTMF does not work.
My topology is:
SBC with PSTN
Server 1 with Asterisk 18.2.2
Server 2 with Asterisk 18.2.2
Server 1 has an pjsip endpoint to SBC and another endpoint to a Server 2.
Server 2 has an pjsip endpoint to Server 1
Basically the call ingress by SBC endpoint in Server 1 and forward to the Server 2.
When direct_media = false on all endpoints de rtp flow between three IPs and de DTMF works fine, but when i configure the direct media = true the rtp flow just between Server 2 and SBC (was expected and is what i want) but the DTMF doesent works.
The DTMF events does not apparece in a log of any server.
The DTMF type to all endpoints is rfc4733.
Below is the configuration of an endpoint:
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (alaw|ulaw|vp8)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : xxxxxxxxx
asymmetric_rtp_codec : false
auth : xxxxxxxxx
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : unknown
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
connected_line_method : invite
contact_acl :
context : from-internal
cos_audio : 0
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_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 : xxxxxxxxx
g726_non_standard : false
ice_support : false
identify_by : username
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : pt_BR
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
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 : xxxxxxxxx
outbound_proxy :
outgoing_call_offer_pref : remote_merge
pickup_group :
preferred_codec_only : false
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 : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
stir_shaken : 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 : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
> DTMF with direct media
> ----------------------
>
> Key: ASTERISK-30439
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-30439
> Project: Asterisk
> Issue Type: Information Request
> Security Level: None
> Components: pjproject/pjsip
> Affects Versions: 18.2.2
> Environment: Docker container
> Alpine Linux
> Asterisk 18.2.2
> Reporter: Vitor Gomes Faria
> Labels: fax, webrtc
>
> Hi guys,
> I'm having some difficulties establishing a topology using direct media.
> The issue is when im using the direct_media = true the DTMF does not work.
> My topology is:
> SBC with PSTN
> Server 1 with Asterisk 18.2.2
> Server 2 with Asterisk 18.2.2
> Server 1 has an pjsip endpoint to SBC and another endpoint to a Server 2.
> Server 2 has an pjsip endpoint to Server 1
> Basically the call ingress by SBC endpoint in Server 1 and forward to the Server 2.
> When direct_media = false on all endpoints de rtp flow between three IPs and de DTMF works fine, but when i configure the direct media = true the rtp flow just between Server 2 and SBC (was expected and is what i want) but the DTMF doesent works.
> The DTMF events does not apparece in a log of any server.
> The DTMF type to all endpoints is rfc4733.
> Below is the configuration of an endpoint:
> ParameterName : ParameterValue
> ===================================================================================================
> 100rel : yes
> accept_multiple_sdp_answers : false
> accountcode :
> acl :
> aggregate_mwi : true
> allow : (alaw|ulaw|vp8)
> allow_overlap : true
> allow_subscribe : true
> allow_transfer : true
> aors : xxxxxxxxx
> asymmetric_rtp_codec : false
> auth : xxxxxxxxx
> bind_rtp_to_media_address : false
> bundle : false
> call_group :
> callerid : unknown
> callerid_privacy : allowed_not_screened
> callerid_tag :
> codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
> codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
> codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
> codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
> connected_line_method : invite
> contact_acl :
> context : from-internal
> cos_audio : 0
> 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_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 : xxxxxxxxx
> g726_non_standard : false
> ice_support : false
> identify_by : username
> ignore_183_without_sdp : false
> inband_progress : false
> incoming_call_offer_pref : local
> incoming_mwi_mailbox :
> language : pt_BR
> mailboxes :
> max_audio_streams : 1
> max_video_streams : 1
> media_address :
> media_encryption : no
> media_encryption_optimistic : false
> media_use_received_transport : false
> 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 : xxxxxxxxx
> outbound_proxy :
> outgoing_call_offer_pref : remote_merge
> pickup_group :
> preferred_codec_only : false
> 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 : true
> rtp_timeout : 0
> rtp_timeout_hold : 0
> sdp_owner : -
> sdp_session : Asterisk
> send_connected_line : yes
> send_diversion : true
> send_history_info : false
> send_pai : false
> send_rpid : false
> set_var :
> srtp_tag_32 : false
> stir_shaken : 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 : false
> use_avpf : false
> use_ptime : false
> user_eq_phone : false
> voicemail_extension :
> webrtc : no
> Did I forget some setting for this to work?
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