[asterisk-bugs] [JIRA] (ASTERISK-16799) Callee declined when 'beep' audio file does not exist

Thu Mar 11 11:48:18 CST 2021

     [ https://issues.asterisk.org/jira/browse/ASTERISK-16799?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-16799:
-------------------------------------

    Target Release Version/s: 18.3.0

> Callee declined when 'beep' audio file does not exist
> -----------------------------------------------------
>
>                 Key: ASTERISK-16799
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-16799
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Applications/app_page
>            Reporter: IAMJames_
>      Target Release: 16.17.0, 18.3.0
>
>
> Assuming this is not the intended behavior,
> The 'Callee' of a page will receive a SIP DECLINE upon issuing the Page command when the 'beep' audio file does not exist.
> ****** ADDITIONAL INFORMATION ******
> <--- SIP read from 192.168.100.209:5060 --->
> INVITE sip:888 at 192.168.100.6 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.209:5060;branch=z9hG4bK-14d6c57b
> From: "JE" <sip:JE at 192.168.100.6>;tag=a567d3c38f3c4ff7o0
> To: <sip:888 at 192.168.100.6>
> Call-ID: d03108b3-6c31fc7 at 192.168.100.209
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest <removed>
> Contact: "JE" <sip:JE at 192.168.100.209:5060>
> Expires: 240
> User-Agent: Linksys/SPA942-6.1.5(a)
> Content-Length: 397
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
> Content-Type: application/sdp
> v=0
> o=- 5727 5727 IN IP4 192.168.100.209
> s=-
> c=IN IP4 192.168.100.209
> t=0 0
> m=audio 16452 RTP/AVP 0 2 4 8 18 96 97 98 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> <------------->
> --- (15 headers 18 lines) ---
> Sending to 192.168.100.209 : 5060 (no NAT)
> Using INVITE request as basis request - d03108b3-6c31fc7 at 192.168.100.209
> Found user 'JE'
> Found RTP audio format 0
> Found RTP audio format 2
> Found RTP audio format 4
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 96
> Found RTP audio format 97
> Found RTP audio format 98
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format G726-32 for ID 2
> Found audio description format G723 for ID 4
> Found audio description format PCMA for ID 8
> Found audio description format G729a for ID 18
> Found unknown media description format G726-40 for ID 96
> Found unknown media description format G726-24 for ID 97
> Found unknown media description format G726-16 for ID 98
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x4040e (gsm|ulaw|alaw|ilbc|h261), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 192.168.100.209:16452
> Looking for 888 in international (domain 192.168.100.6)
> list_route: hop: <sip:JE at 192.168.100.209:5060>
> <--- Transmitting (no NAT) to 192.168.100.209:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.100.209:5060;branch=z9hG4bK-14d6c57b;received=192.168.100.209
> From: "JE" <sip:JE at 192.168.100.6>;tag=a567d3c38f3c4ff7o0
> To: <sip:888 at 192.168.100.6>
> Call-ID: d03108b3-6c31fc7 at 192.168.100.209
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:888 at 192.168.100.6>
> Content-Length: 0
> <------------>
>     -- Executing [888 at international:1] NoOp("SIP/JE-00000dcd", "All staff page from 888") in new stack
>     -- Executing [888 at international:2] Page("SIP/JE-00000dcd", "local/test") in new stack
>     -- Called test
>     -- Executing [test at default:1] Answer("Local/test at default-1a25,2", "") in new stack
>     -- Executing [test at default:2] Playback("Local/test at default-1a25,2", "vm-delete") in new stack
>     -- <Local/test at default-1a25,2> Playing 'vm-delete' (language 'en')
>   == Spawn extension (international, 888, 2) exited non-zero on 'SIP/JE-00000dcd'
> Scheduling destruction of SIP dialog 'd03108b3-6c31fc7 at 192.168.100.209' in 32000 ms (Method: INVITE)
> <--- Reliably Transmitting (no NAT) to 192.168.100.209:5060 --->
> SIP/2.0 603 Declined
> Via: SIP/2.0/UDP 192.168.100.209:5060;branch=z9hG4bK-14d6c57b;received=192.168.100.209
> From: "JE" <sip:JE at 192.168.100.6>;tag=a567d3c38f3c4ff7o0
> To: <sip:888 at 192.168.100.6>;tag=as672a5c25
> Call-ID: d03108b3-6c31fc7 at 192.168.100.209
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
> <------------>
>   == Spawn extension (default, test, 2) exited non-zero on 'Local/test at default-1a25,2'
> <--- SIP read from 192.168.100.209:5060 --->
> ACK sip:888 at 192.168.100.6 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.209:5060;branch=z9hG4bK-14d6c57b
> From: "JE" <sip:JE at 192.168.100.6>;tag=a567d3c38f3c4ff7o0
> To: <sip:888 at 192.168.100.6>;tag=as672a5c25
> Call-ID: d03108b3-6c31fc7 at 192.168.100.209
> CSeq: 102 ACK
> Max-Forwards: 70
> Proxy-Authorization: <removed>
> Contact: "JE" <sip:JE at 192.168.100.209:5060>
> User-Agent: Linksys/SPA942-6.1.5(a)
> Content-Length: 0


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