| Summary: | ASTERISK-26158: Asterisk 13.1-cert1 : no RTP when using local channels and sip channel in extensions.conf | ||
| Reporter: | NOC Afone (noc_level3) | Labels: | |
| Date Opened: | 2016-06-28 08:11:24 | Date Closed: | 2020-01-14 11:13:49.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Bridges/bridge_native_rtp Channels/chan_bridge |
| Versions: | 13.1.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | CentOS release 5.6 (Final) | Attachments: | ( 0) debug_log_123456_ko |
| Description: | 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 (0111111111) here is an extract of extensions.conf -------------------------------------------------------------- exten => _0170131121,1,Progress() exten => _0170131121,n,Dial(Local/S00111111111@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@afone.com AFONE | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-06-28 08:11:25.560-0500 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. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: NOC Afone (noc_level3) 2016-06-28 08:12:44.946-0500 log and debug file By: Joshua C. Colp (jcolp) 2016-06-28 08:22:44.862-0500 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. By: Asterisk Team (asteriskteam) 2016-07-12 12:00:01.029-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||