[asterisk-bugs] [JIRA] (ASTERISK-29403) Asterisk can't bridge or detect the second call leg without Answer()ing on early media.

Stanislav Abramenkov (JIRA) noreply at issues.asterisk.org
Fri Apr 23 01:10:09 CDT 2021


Stanislav Abramenkov created ASTERISK-29403:
-----------------------------------------------

             Summary: Asterisk can't bridge or detect the second call leg without Answer()ing on early media.
                 Key: ASTERISK-29403
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29403
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Bridges/bridge_interval_features
    Affects Versions: 16.17.0
         Environment: Debian 10.9
            Reporter: Stanislav Abramenkov


Shortly:
Asterisk can't bridge or detect the second call leg without Answer()ing on early media.

I will try to describe the problem in more detail. Seems that asterisk 16.17.0 have issue with detecting early media. 

I have two asterisk servers.
One of server is running old asterisk version - 11.7.0
And second server has version - 16.17.0

Both servers have connection to Oracle SBC.
Description of call flow:
asterisk (Public IP) ==> SBC (Public IP) ==> other system

When the call made from asterisk 11.7.0 to SBC, then SBC will send this call to other system and if extension not found, then other system plays notification (early media) caller in asterisk hears it, so first and second call legs are properly connected. (picture "pic1" in attachment)

On new system with asterisk 16.17.0, it doesn't happen, caller only hears dialing tone, but in trace (tcpdump) we see early media notification. And RTP of early media can be played in wireshark. (picture "pic2" in attachment)

SIP trunk configurations are same on both servers:

[sbctrunk1]
description=earlym
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
allow=alaw
allow=g722
host=public_ip
permit=public_ip
transport=tcp,udp
port=5060
qualifyfreq=60
qualify=3000
canreinvite=no
dtmfmode=auto
progressinband=never
nat=no
directrtpsetup=no
directmedia=no
context=sbctrunk1
insecure=port,invite
promiscredir=yes
accountcode=sbctrunk1

Dial plan:

exten => _X.,1,NoOP(TEST)
exten => _X.,n,Dial(SIP/${EXTEN}@sbctrunk1,,)
exten => _X.,n,Hangup()



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