[asterisk-bugs] [JIRA] (ASTERISK-21845) maxcalls exceeded, Asterisk sends out 480 and also BYE

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-21845?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-21845:
-------------------------------------

    Target Release Version/s: 14.0.0

> maxcalls exceeded, Asterisk sends out 480 and also BYE
> ------------------------------------------------------
>
>                 Key: ASTERISK-21845
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21845
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.22.0
>         Environment: CentOS 5.5 64-bits
>            Reporter: Tony Ching
>            Severity: Minor
>      Target Release: 14.0.0
>
>
> maxcalls in asterisk.conf was set to limit the number of handling sip call.
> The bug is : Asterisk sends 480 Temporarily Unavailable to the incoming sip call, after reaching the number of maxcalls (which is ok). The SIP client sends ACK (this is normal). Then, Asterisk sends a BYE (which is not correct).
> == pcap =======
> Time abs Source Destination Protocol Length Info
> 0.0 192.168.3.79 192.168.3.217 SIP/SDP 824 Request: INVITE sip:513 at 192.168.3.217:6061, with session description
> 0.0 192.168.3.217 192.168.3.79 SIP 544 Status: 100 Trying
> 0.0 192.168.3.217 192.168.3.79 SIP 537 Status: 480 Temporarily Unavailable
> 0.0 192.168.3.79 192.168.3.217 SIP 354 Request: ACK sip:513 at 192.168.3.217:6061
> 0.0 192.168.3.217 192.168.3.79 SIP 464 Request: BYE sip:1234567 at 192.168.3.79:6763
> 0.0 192.168.3.79 192.168.3.217 SIP 361 Status: 481 Call/Transaction Does Not Exist
> == ast debug output =======
> Maximum call limit of 1 calls exceeded by 'SIP/sip_incoming-00000002'!
> Failed to start PBX (call limit reached)
> Trying to put 'SIP/2.0 480' onto UDP socket destined for 192.168.3.79:6763
> Hanging up channel 'SIP/sip_incoming-00000002'
> Hangup call SIP/sip_incoming-00000002, SIP callid 954837550a18de12 at YWxleGNoYW4tcGM.
> Setting RTCP address on RTP instance '0x10bfe478'
> No provider found, checking channel drivers for SIP - sip_incoming
> Checking device state for peer sip_incoming
> Changing state for SIP/sip_incoming - state 1 (Not in use)
> device 'SIP/sip_incoming' state '1'
> No provider found, checking channel drivers for SIP - sip_incoming
> Checking device state for peer sip_incoming
> Changing state for SIP/sip_incoming - state 1 (Not in use)
> device 'SIP/sip_incoming' state '1'
> Setting the marker bit due to a source update
> Setting the marker bit due to a source update
> Setting the marker bit due to a source update
> **** Received ACK (devil) - Command in SIP ACK
> Stopping retransmission on '954837550a18de12 at YWxleGNoYW4tcGM.' of Response 1: Match Found
> Trying to put 'BYE sip:852' onto UDP socket destined for 192.168.3.79:6763



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