similar to: Asterisk 1.6.2.6 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.0.26. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.26 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.0.26. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.26 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

A buggy or malicious frontend should not be able to confuse netback. If we spot anything which is not as it should be then shutdown the device and don''t try to continue with the ring in a potentially hostile state. Well behaved and non-hostile frontends will not be penalised. As well as making the existing checks for such errors fatal also add a new check that ensures that there

VM traffic is already limited by a throughput limit, but there is no control over the maximum packet per second (PPS). In DDOS attack the major issue is rather PPS than throughput. With provider offering more bandwidth to VMs, it becames easy to coordinate a massive attack using VMs. Example: 100Mbits ~ 200kpps using 64B packets. This patch provides a new option to limit VMs maximum packets per

Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

(sorry this is so long) I could really use a helping hand. I have a 1.6.2.6 installation using LDAP as the realtime engine for voicemail users, SIP users, queues, and some custom hotdesking families. I'm also using ODBC voicemail storage. The issue that I am having is that the UA's (Polycom 501's, 301's and 430's) are receiving multiple SIP NOTIFY messages when the user

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom

I've just upgraded to 1.6.2.6 on one of my test systems. I started out happy, with some improvements in transfers to Local() channels from a SIP channel, and much nicer verbose fax handling. However, something is really weird when I need to do System() calls. It was really, really weird. This was also affecting AGI, when I needed to read system variables from asterisk into an AGI Perl script.

hi list, after connecting 3 asterisk servers via IAX in a line (+ 1 client at each end), i noticed that call path optimization happens only one time, i.e. only one node/leg in the path can be reduced. Does anybody know if this is the intended behaviour or if it's a bug? Can anyone confirm my observations? It seems that the first node, that sends the TXREQ Message, is "optimized".

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

Another checksum offload problem was reported on xen-users, when using a domU as a firewall: http://lists.xensource.com/archives/html/xen-users/2006-04/msg01150.html It also fails without VLANs. The path from dom0->domU with ip_summed==CHECKSUM_HW/proto_csum_blank==1 is broken. - skb_checksum_setup() assumes that a checksum will definitely be calculated in dev_queue_xmit(), but the

Hello, I have the same problem, you said that was a configuration problem. I play with configuration but can not get it working. Can you please show you working configuration. Any help appreciated. -------------------------------------------original messages----------------------------------------------------------------------------------- Hi, it was some configuration error. I droped the

Hello, Anyone have any insight or fix for the error below? It was the last error in the log before Asterisk crashed. I am running Asterisk 1.6.2.6 only for the T.38 support. 06:21:49] ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758 Google has some vague references that there is a patch or something but nothing I could really follow. Anyone have any ideas? The system was up for about