[asterisk-bugs] [JIRA] (ASTERISK-23637) Missinterpretation of DTMF from GXP-2200 in res_rtp_asterisk.c

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23637?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217816#comment-217816 ] 

Rusty Newton commented on ASTERISK-23637:
-----------------------------------------

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines



> Missinterpretation of DTMF from GXP-2200 in res_rtp_asterisk.c
> --------------------------------------------------------------
>
>                 Key: ASTERISK-23637
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23637
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.8.1
>         Environment: Ubuntu 13.04 
> GXP-2200 FW 1.0.3.26
> Asterisk 11.8.1 and Asterisk 1.8.26.1
>            Reporter: Olaf Winkler
>            Assignee: Michael L. Young
>            Severity: Minor
>
> When sendig DTMF-Digits (rfc2833, here: feturecode ## for blind transfer) from an Granstream GXP-2200 these tones are missinterpreted in Asterisk 11.8.1
> {quote}
> [Apr 15 09:39:59] VERBOSE[10145][C-000003b9] pbx.c:     -- Executing [XXXXXXX at sip_phone:36] Dial("SIP/13-000003ae", "Capi/g1/XXXXXXX/b,30,ciHT") in new stack
> [Apr 15 09:39:59] VERBOSE[10145][C-000003b9] app_dial.c:     -- Called Capi/g1/XXXXXXX/b
> [Apr 15 09:40:00] VERBOSE[10145][C-000003b9] app_dial.c:     -- CAPI/ISDN1#02/XXXXXXX-6b4 is proceeding passing it to SIP/13-000003ae
> [Apr 15 09:40:01] VERBOSE[10145][C-000003b9] app_dial.c:     -- CAPI/ISDN1#02/XXXXXXX-6b4 is making progress passing it to SIP/13-000003ae
> [Apr 15 09:40:01] VERBOSE[10145][C-000003b9] app_dial.c:     -- CAPI/ISDN1#02/XXXXXXX-6b4 is ringing
> [Apr 15 09:40:01] VERBOSE[10145][C-000003b9] res_rtp_asterisk.c:        > 0xb6f19b78 -- Probation passed - setting RTP source address to XXX.XXX.XXX.XXX:XXXXX
> [Apr 15 09:40:03] VERBOSE[10145][C-000003b9] app_dial.c:     -- CAPI/ISDN1#02/XXXXXXX-6b4 answered SIP/13-000003ae
> [Apr 15 09:40:15] VERBOSE[10145][C-000003b9] res_musiconhold.c:     -- Started music on hold, class 'default', on CAPI/ISDN1#02/XXXXXXX-6b4
> [Apr 15 09:40:35] ERROR[10145][C-000003b9] res_rtp_asterisk.c: Received SSL traffic on RTP instance '0xb6f1566c' without an SSL session
> [Apr 15 09:40:35] WARNING[10145][C-000003b9] res_rtp_asterisk.c: RTP Read error: Unspecified.  Hanging up.
> [Apr 15 09:40:35] VERBOSE[10145][C-000003b9] pbx.c:     -- Executing [h at sip_phone:1] Hangup("SIP/13-000003ae", "") in new stack
> {quote}
> res_rtp_asterisk interprets the DIGITS as SSL-Traffic which is not correct.
> When testing the same situation with Asterisk 1.8.26.1 the same telephone behalfs normal - the DTMF-digits are interpreted correctly by Asterisk.
> So something seems to be wrong in the packet interpretation in Asterisk 11.
> Unfortunately I'm not able to reproduce this error in my own environment as I found it in a customer ones where I'm not able to execute phone-tests from remote.



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