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

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23637?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217371#comment-217371 ] 

Michael L. Young commented on ASTERISK-23637:
---------------------------------------------

Olaf,  Can you get a PCAP at all and attach it?

Asterisk 1.8 did not have DTLS-SRTP in it and therefore whatever is coming from the Grandstream was probably being ignored.  For 11, whatever is coming from the Grandstream is now being picked up by Asterisk.  It would be helpful to see what the Grandstream is sending.

> 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
>            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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)