| Summary: | ASTERISK-21845: maxcalls exceeded, Asterisk sends out 480 and also BYE | ||||
| Reporter: | Tony Ching (tonyching) | Labels: | |||
| Date Opened: | 2013-05-29 21:25:53 | Date Closed: | 2013-05-31 20:48:13 | ||
| Priority: | Minor | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | 1.8.22.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | CentOS 5.5 64-bits | Attachments: | |||
| Description: | 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@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@192.168.3.217:6061 0.0 192.168.3.217 192.168.3.79 SIP 464 Request: BYE sip:1234567@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@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@YWxleGNoYW4tcGM.' of Response 1: Match Found Trying to put 'BYE sip:852' onto UDP socket destined for 192.168.3.79:6763 | ||||
| Comments: | By: Rusty Newton (rnewton) 2013-05-31 20:48:13.470-0500 Thanks for the report. Closing as duplicate of ASTERISK-15434 By: Rusty Newton (rnewton) 2013-05-31 20:50:03.408-0500 There is a patch on ASTERISK-15434. However it is old, so theres a chance you may have to modify it. | ||||