The Asterisk Development Team has announced the release of Asterisk 10.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.6.0 resolves several issues reported by the community like:
- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)
- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis)
- Re-add LastMsgsSent value for SIP peers (Closes issue ASTERISK-17866. Reported by Steve Davies)
- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. (Closes issue ASTERISK-19425. Reported by David Cunningham)
- Send more accurate identification information in dialog-info SIP NOTIFYs. (Closes issue ASTERISK-16735.…
The Asterisk Development Team has announced the release of Asterisk 18.104.22.168. This release is available for immediate download at ttp://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 22.214.171.124 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) * --- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis) * --- Update a peer's LastMsgsSent when the peer is notified of waiting…
Asterisk Project Security Advisory - AST-2012-010
Product Asterisk Summary Possible resource leak on uncompleted re-invite transactions Nature of Advisory Denial of Service Susceptibility Remote authenticated sessions Severity Minor Exploits Known No Reported On June 13, 2012 Reported By Steve Davies Posted On July 5, 2012 Last Updated On July 5, 2012 Advisory Contact Terry Wilson
Description If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for…
*Bump* No takers? Perhaps no-one else thinks this is a bug?
Steve On 7 February 2011 16:45, Steve Davies
> The following IAX config (slightly edited) causes an issue for me in
> version 126.96.36.199.1, where my CDR data is unreliable.
On 09/09/10 17:59, Steve Davies wrote:
> On 9 September 2010 17:52, Antonio Berrios
>> Steve Davies wrote:
>>> I am using 188.8.131.52, and I need to be able to include the name of the
>>> channel that answered a call in the call-recording filename.
>>> At a guess we need to use the Queue(name,,,,,,macro) or
>>> Dial(chan1&chan2,,M(macro)) and use the macro to update the call
>>> recording filename. But, the macro runs on the calling channel, and…