Re: [asterisk-users] Asterisk 1.8.0-beta4 Now Available
At 08:35 AM 8/24/2010, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4. I've now tried all the V1.8 betas including this and I always get a message telling me to read sip-retransmit.txt when I make a call from a SIP phone, Aastra480i out a DAHDI line on a Digium TDM-400 with 4 red cards back to one of the other lines. It rings once and then I get 4 or so of that message and it goes to voicemail. Soon as I go back to the latest 1.6 it works perfectly again. I've read the document many time and I have no clue what to do with the information. I only have one box to test on so I just test it the occasional quiet evening by making that one call and it always fails with these message. I have no idea what to do to try and make it work or if it's likely a bug or an error in my configuration. I got no errors in loading that should have any effect on this. So this is a show stopper for me and while I'd love to help test 1.8, I can't successfully make one call with it up. Any suggestions on what I might try to improve things? [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 3038c0be7937f81a5a5187441e4b3...@192.168.2.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Hanging up call 3038c0be7937f81a5a5187441e4b3...@192.168.3.235:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 4427982a1e30f2e06aded749152d4...@192.168.3.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Hanging up call 4427982a1e30f2e06aded749152d4...@192.168.3.235:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). [2010-09-03 23:51:46] WARNING[29354] chan_sip.c: Retransmission timeout reached on transmission 0226b2630404b2eb3aa8a5eb789e9...@192.168.3.235:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.0-beta4 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page. http://www.asterisk.org/asterisk-versions This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include: * Fix parsing of IPv6 address literals in outboundproxy (Closes issue #17757. Reported by oej. Patched by sperreault) * Change the default value for alwaysauthreject in sip.conf to yes. (Closes issue #17756. Reported by oej) * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip. (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/) * PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. (Patched by rmudgett) * Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode. (Patched by russellb) * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list the special CEL variables that are available for use in the mapping. There are also several other CEL fixes in this release. (Patched by russellb) Asterisk 1.8 contains many new features over previous releases of Asterisk. A short list of included features includes: * Secure RTP * IPv6 Support in the SIP Channel * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta4 Now Available
Hi! * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip. (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/) What you do not see mentioned here, and that is a bit misleading: Instead res_stun_monitor was added to be able to detect dynamic IP changes. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
