[asterisk-bugs] [JIRA] (ASTERISK-25379) no sound on pjsip channel with bridge_native_rtp enabled
Thomas Stein (JIRA)
noreply at issues.asterisk.org
Sun Sep 6 08:41:33 CDT 2015
Thomas Stein created ASTERISK-25379:
---------------------------------------
Summary: no sound on pjsip channel with bridge_native_rtp enabled
Key: ASTERISK-25379
URL: https://issues.asterisk.org/jira/browse/ASTERISK-25379
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_pjsip
Affects Versions: 13.5.0
Environment: Gentoo Linux, Kernel 4.2.0.
Reporter: Thomas Stein
Severity: Critical
Hello.
I'm getting no sound out of my phone (pjsip endoint) while bridge_native_rtp is enabled. As soon as i disable the function in menuconfig everything works as expected.
Here is my enpoint info:
ParameterName : ParameterValue
====================================================
100rel : yes
accountcode :
aggregate_mwi : true
allow : (ulaw|g722|gsm)
allow_subscribe : true
allow_transfer : true
aors : 501
auth : auth501
call_group :
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
context : 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 : true
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
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username
inband_progress : false
language : en
mailboxes :
media_address :
media_encryption : no
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 : true
rpid_immediate : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 3
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : false
send_rpid : false
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 : 0
tos_video : 0
transport : transport-udp
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
I can provide more information if needed.
thanks and cheers
t.
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