FreePBX | Register | Issues | Wiki | Portal | Support

Freepbx14 Videocall not work

configuration
asterisk
freepbx
Tags: #<Tag:0x00007f7499f736a8> #<Tag:0x00007f7499f73478> #<Tag:0x00007f7499f73338>

(Otaro) #1

Hello

             today i test  video call  is not work     please help me

i enabled Video Support and select codec success but not work

This is log

– Called PJSIP/101/sip:101@1XX.1XX.XX.XXX:61780
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for codec2
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
– Connected line update to PJSIP/102-00000014 prevented.
– PJSIP/101-00000015 is ringing
– PJSIP/101-00000015 answered PJSIP/102-00000014
– Channel PJSIP/101-00000015 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/102-00000014 joined ‘simple_bridge’ basic-bridge
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
== Using SIP RTP Video CoS mark 4
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
== Using SIP RTP Video CoS mark 4

Thankyou


(Otaro) #2

Please Help me


(Dave Burgess) #3

Not without logs. There are literally a hundred things that could be going wrong and without some kind of clues, it’s just a waste of everyone’s time.


(Otaro) #4

This is log

– Called PJSIP/101/sip:101@1XX.1XX.XX.XXX:61780
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for codec2
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[2019-05-08 11:23:09] WARNING[32697]: res_pjsip_sdp_rtp.c:1514 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
– Connected line update to PJSIP/102-00000014 prevented.
– PJSIP/101-00000015 is ringing
– PJSIP/101-00000015 answered PJSIP/102-00000014
– Channel PJSIP/101-00000015 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/102-00000014 joined ‘simple_bridge’ basic-bridge
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
== Using SIP RTP Video CoS mark 4
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
== Using SIP RTP Video CoS mark 4


(system) closed #5

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.