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This is the NEWS for chan_ss7. This version is maintained by Anders Baekgaard (ab@dicea.dk).Please send bug reports, feature requests etc. to chan_ss7@dicea.dk.New in version 1.1- Fixed buffer overflows in config.c- Fixed loss of IDLE CICs- Fixed segmentation fault when using "combined" attribute for linksets.- Fixed block/unblock of last cic not possible bug- Fixed handling of dial request supporting multiple audio formats- Support for STP signalling, see file ss7.conf.template.single-link for config- Jitter buffer handling (thanks to Martin V韙, sponsored by www.voipex.cz)- H324M support (thank to Klaus Darilion)- Fixed a bug that could cause one-way audio in some cases where DTMF codes are sent.- Fixed a bug where receive fifo is no longer being readNew in verion 1.0.0- Compatible with asterisk 1.2.x and 1.4.x.- MTP stack placed in standalone executable.- New loadshare config parameter for linksets (None, linkset, combined).- New combined config parameter for linksets. Linksets having the same combined setting and having loadshare=combined share signalling channels.- New auto_block config parameter for links. When set to yes, the CICs on that link are blocked when signalling on the link is lost.- The schannel entry in link description in configuration file may specify remote MTP stack.- PDU dump is now in PCAP format, suitable for wireshark.- Lots and lots of clean ups and fixes.
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