[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 06:42:09 CDT 2021
[ https://issues.asterisk.org/jira/browse/ASTERISK-29403?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Stanislav Abramenkov updated ASTERISK-29403:
--------------------------------------------
Attachment: pic2.jpeg
pic1.jpeg
call flow
> 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: Channels/chan_sip/General, Resources/res_rtp_asterisk
> Affects Versions: 16.17.0
> Environment: Debian 10.9
> Reporter: Stanislav Abramenkov
> Assignee: Unassigned
> Attachments: ast_11_7_0_dump.txt, ast_16_17_0_dump.txt, pic1.jpeg, pic2.jpeg
>
>
> 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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