Unknown RTP codec 95 and Forbidden

Asterisk Asterisk Support
1

Hi,

New here and this is my first post.

I have recently set up a freepbx system and everything seemed to be working fine. I connected an xlite softphone to test the system. I can place calls to mobiles through the system fine but when I try and call any landline numbers I receive the following message.

    -- Called SIP/myprovider/022001441417743000
  == Begin MixMonitor Recording SIP/108-00000005
[2016-09-02 10:53:54] WARNING[45965][C-00000003]: chan_sip.c:23294 handle_response_invite: Received response: "Forbidden" from '"108" <sip:108@"myserver">;tag=as51f0c820'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/108-00000005", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/108-00000005", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/108-00000005", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/108-00000005", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/108-00000005", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/108-00000005", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] Set("SIP/108-00000005", "CALLERID(number)=108") in new stack
    -- Executing [01417743000@from-internal:6] Macro("SIP/108-00000005", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/108-00000005", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/108-00000005", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/108-00000005", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/108-00000005", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/108-00000005> Playing 'all-circuits-busy-now.gsm' (language 'en')
       > 0x7f1ed40634e0 -- Probation passed - setting RTP source address to xxx.xxx.xxx.xxx(myip):60452
[2016-09-02 10:53:54] NOTICE[47106][C-00000003]: res_rtp_asterisk.c:4487 ast_rtp_read: Unknown RTP codec 95 received from 'xxx.xxx.xxx.xxx(myip):60452'
    -- <SIP/108-00000005> Playing 'pls-try-call-later.gsm' (language 'en')
  == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/108-00000005' in macro 'outisbusy'
  == Spawn extension (from-internal, 01417743000, 6) exited non-zero on 'SIP/108-00000005'
    -- Executing [h@from-internal:1] Hangup("SIP/108-00000005", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/108-00000005'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/108-0000000

The RTP codec 95 is the most confusing bit. can’t really find anything about it when googling.

Thanks

2

Forbidden means the peer was capable of doing what you wanted but wasn’t prepared to do it for you, e.g. you provided the wrong password, or weren’t allowed to call a particular number. You need to ask the person operating that peer why it rejected your request.

I had no trouble finding codec 95 issues on Google. I’d suggest: Res_rtp_asterisk.c: Unknown RTP codec 95 received from

Unfortunately, that thread dried up before enough information had been provided to work out what was going wrong.

3

[quote=“david551, post:2, topic:67956”]
I had no trouble finding codec 95 issues on Google. I’d suggest: Res_rtp_asterisk.c: Unknown RTP codec 95 received from [/quote]

Asterisk is complaining that someone sent it an RTP packet with a non-negotiated payload type, which it then had to drop. In order to know which codec this is, you’d have to look at the negotiated SDP.

(99% of the time this is a comfort noise packet)