[asterisk-bugs] [JIRA] (ASTERISK-26158) Asterisk 13.1-cert1 : no RTP when using local channels and sip channel in extensions.conf

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26158?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231220#comment-231220 ] 

Asterisk Team commented on ASTERISK-26158:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

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> Asterisk 13.1-cert1 : no RTP when using  local channels and sip channel in extensions.conf
> ------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26158
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26158
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Channels/chan_bridge
>    Affects Versions: 13.1.0
>         Environment: CentOS release 5.6 (Final)
>            Reporter: NOC Afone
>
> hello,
> We see no RTP packets out of Asterisk when using  local channels and sip channel in that configuration :
> The call flow of the call is : 
> => PSTN (0170131121) => ASTERISK => PSTN (0637286210)
> here is an extract of extensions.conf
> --------------------------------------------------------------
> exten => _0170131121,1,Progress()
> exten => _0170131121,n,Dial(Local/S00637286210 at appelsortant/n,20)
> [appelsortant]
> exten => _S.,1,Set(NETWORKSTATUS=${SIPPEER(STD1-BCT1-VIP-MGC,status)})
> exten => _S.,n,Set(DATACENTER=${IF($["${NETWORKSTATUS}"="UNREACHABLE"]?CBV2:STD1)})
> exten => _S.,n,Dial(SIP/${EXTEN:2}@${DATACENTER}-BCT1-VIP-MGC)
> exten => h,n,Set(SHARED(AF_${NUMDEST}_DIALSTATUS,${AFPARENTCHANNEL})=${CDR(disposition)})
> exten => h,n,Set(SHARED(AF_${NUMDEST}_HANGUPCAUSE,${AFPARENTCHANNEL})=${HANGUPCAUSE})
> exten => h,n,Set(SHARED(AF_${NUMDEST}_DURATION,${AFPARENTCHANNEL})=${CDR(duration)})
> exten => h,n,Set(SHARED(AF_${NUMDEST}_BILLSEC,${AFPARENTCHANNEL})=${CDR(billsec)})
> exten => h,n,Set(SHARED(AF_${NUMDEST}_COMPLETE,${AFPARENTCHANNEL})=1)
> --------------------------------------------------------------
> A tcpdump capture shows there are no RTP packets out of ASTERISK.
> -----
> We have no problems :
> -  When we replace
> exten => _S.,n,Dial(SIP/${EXTEN:2}@${DATACENTER}-BCT1-VIP-MGC)
> by
> exten => _S.,n,Dial(SIP/${EXTEN:2}@${DATACENTER}-BCT1-VIP-MGC,,r) 
> We force ASTERISK to generate in ringbacktone : we see rtp packets out of Asterisk.
> - when we decide to play an audio file before Dial(SIP/${EXTEN:2}@${DATACENTER}-BCT1-VIP-MGC).
> Regards
> Abdoul OSSENI
> aosseni at afone.com
> AFONE



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