similar to: Asterisk 1.6.1.18 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.2.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.6 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 community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.6 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 community

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

Hello, When I run the 'smbclient -L apollo' command at the UNIX prompt a I get an error. I have gone as far as I can in this DIAGNOSIS.txt file. Does anyone have any thoughts on what may be causing my problem ? I have made my comments to the DIAGNOSIS.txt file via '---- davidw {comment}' so you can see what I've done to date. Also, I thought I'd pass on the output

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

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0. All interested users of Asterisk are encouraged to

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0. All interested users of Asterisk are encouraged to

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

Hi, I''m running a more or less up-to-date system, and am encountering something that''s irritating me: I make changes to the .rhtml file, and yet don''t see them from my browser. The development.log file indicates that 1) I am indeed in development mode, and 2) queries are being run against the database, so something is happening - it''s just not reading in the

Hi, I''m trying to get rails running on Debian stable with Apache 2, mod_fcgid, Rails 1.0. The application works fine with webrick (*). Let''s start here: @eugene [/var/www/ls2/linuxsi/public] $ ./dispatch.fcgi Status: 500 Internal Server Error Seems to result in the following in the log files: [12/Feb/2006:12:18:40 :: 4105] starting [12/Feb/2006:12:18:40 :: 4105] terminated

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'm running 1.6.1.18 on an older ubuntu machine. I upgraded to dahi-linux-2.3.0. That went fine, and it works. But I decided to use the opportunity to upgrade to 1.6.2.6. That didn't work. configure, make menuselect, make, make install all went fine, or at least seemed to. But it hangs starting up here: [Apr 13 20:15:28] VERBOSE[1612] codec_speex.c: -- CODEC SPEEX: Setting

Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [s at fax-tx-test:3] SendFAX("SIP/side-sip-00000009", "/var/spool/asterisk/fax/20091113_1455.tif") in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped