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

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26158?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-26158:
-----------------------------------

    Assignee: NOC Afone
      Status: Waiting for Feedback  (was: Triage)

The certified series only receives fixes as a result of issues encountered by commercial customers. If you are a commercial customer please submit a support ticket to Digium so it can be handled.

If you are not a commercial customer you will need to use the latest version of 13 before we are able to look at this issue and provid updated logging. As well a better description of the network layout (such as if NAT is involved at all) and wireshark capture will be needed.

> 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
>            Assignee: NOC Afone
>         Attachments: debug_log_123456_ko
>
>
> 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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