[asterisk-bugs] [JIRA] (ASTERISK-17044) Call torn down upon connection when early media 183 used

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-17044?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-17044.
----------------------------------

    Resolution: Fixed

Closing as "Fixed" per the last comment.

If this problem is still an issue in Asterisk, please contact a bug marshal in #asterisk-bugs or comment on this issue and we'd be happy to reopen it. Thanks!

> Call torn down upon connection when early media 183 used
> --------------------------------------------------------
>
>                 Key: ASTERISK-17044
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17044
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/General
>            Reporter: Erik Smith
>         Attachments: GateWayDebugLog, gw-to-carrier-8006262001-1.pcap, gw-to-carrier-8006262001-2.pcap, issue_18399_full_log.txt, me-to-18006262001.pcap, phone-endpoint-18006262001.pcap
>
>
> Asterisk 1.8.1-rc1 & Asterisk 1.6.2.14
> Centos 5.5
> have scenario as such
> Asterisk-1.8.1-rc1 -SIP-> Asterisk 1.6.2.14 -SIP-> Broadvox (Sonus Softswitch)
> When calling a TF number that uses early media for their IVR (example 1-800-626-2001); once the call gets connected and the 200 OK message is received, my 1.8.1-rc1 box issues a BYE message with a HangupCauseCode of 0. I can reproduce this with several numbers that are using early media for their IVR's. Just as soon as my call gets connected to a call-center's ACD Queue I hear 1-2 seconds of the recording before the call is torn down. I have tested this using a linksys SPA-2102 ATA, A Polycom IP501, as well as a Digium FXS module and get identical results. 
> ****** ADDITIONAL INFORMATION ******
>     -- Called +18006262001 at GW01EEMAN
>     -- SIP/GW01EEMAN-00000039 is making progress passing it to SIP/202-00000038
> [Nov 29 16:24:26] WARNING[25528]: chan_sip.c:12909 __set_address_from_contact: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
> [Nov 29 16:24:26] ERROR[25528]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("Ã     ", "(null)", ...): Name or service not known
> [Nov 29 16:24:26] WARNING[25528]: chan_sip.c:12920 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'Ã        '
>     -- SIP/GW01EEMAN-00000039 answered SIP/202-00000038
>   == Spawn extension (macro-tl-dialout-base, dial-SIP, 7) exited non-zero on 'SIP/202-00000038' in macro 'tl-dialout-base'
>   == Spawn extension (macro-tl-dialout-1-trunk, s, 3) exited non-zero on 'SIP/202-00000038' in macro 'tl-dialout-1-trunk'
>   == Spawn extension (from-inside-redir, 18006262001, 1) exited non-zero on 'SIP/202-00000038'
>     -- Executing [h at from-inside-redir:1] Hangup("SIP/202-00000038", "") in new stack
>   == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'SIP/202-00000038'



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