Unsolved Sms chat not working on freepbx with tow linephone softphone
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i am using freepbx 15 distro
Other SIP Settings
accept_outofcall_message = yes
outofcall_message_contex = astsms
auth_message_requests = yes
Write below lines in extensions_custom.conf file. This is dialplan to send IM.
[astsms]
;Deliver to local 3-digit extension
exten => _XXX,1,MessageSend(sip:${EXTEN},${MESSAGE(from)})
same => n,Hangup()res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2021-06-16 22:34:12] ERROR[22726][C-00000005]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 83105471079628703765225689606832e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 10635857318019912468722712362a17e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 177607474807695456872236a3608968e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6402ms with no response
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 163608105081901038768722bc289057e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 38370651581339966687222681d658b7e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 92561586182471080867122238561512e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response -
sip_general_custom.conf
accept_outofcall_message=yes
outofcall_message_context=dialplan_name
auth_message_requests=yes
…sip_general_additional.conf
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
useragent=FPBX-15.0.17.34(17.9.3)
language=en
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
allow=g722
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
accept_outofcall_message=yes
outofcall_message_context=astsms
auth_message_requests=yes
context=from-sip-external
callerid=Unknown
tcpenable=no
callevents=yes
jbenable=no
checkmwi=10
maxexpiry=3600
minexpiry=60
srvlookup=no
tlsenable=no
allowguest=yes
notifyhold=yes
rtptimeout=30
canreinvite=no
tlsbindaddr=[::]:5161
rtpkeepalive=0
videosupport=no
defaultexpiry=120
notifyringing=yes
maxcallbitrate=384
rtpholdtimeout=300
g726nonstandard=no
registertimeout=20
tlsclientmethod=tlsv1
registerattempts=0
nat=force_rport,comedia
ALLOW_SIP_ANON=no
udpbindaddr=0.0.0.0:5060
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
externip=104.145.12.182
localnet=192.168.1.6/24
…sip_additional.conf
[100]
deny=0.0.0.0/0.0.0.0
secret=12345
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/100
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Omer RIT <100>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no[101]
deny=0.0.0.0/0.0.0.0
secret=123
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/101
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Ahmed RIT <101>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
