similar to: [Bug 498] New: RTP packets are not hitting NAT table

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 cfilin@intermedia.net changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |chip@innovates.com -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are

The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from "A" to "B" (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name

Hi Quentin, On 08/19/14 18:58, Quentin Colombet wrote: [...] > It seems that you will have to debug further the *** Bad machine code: Instruction loads from dead spill slot *** before we can be of any help. Yes, I've done some more digging. Sorry for the long mail... I get: Inline spilling aN40_0_7:%vreg1954 [5000r,5056r:0)[5056r,5348r:1) 0 at 5000r 1 at 5056r At this point I have

Hi Kotresh, I am expecting my hardware to show up next week. My plan is to run gluster version 3.12 on centos 7. Has the issue been fixed in version 3.12? Thanks a lot for your help! /Marcus On Fri, Mar 02, 2018 at 05:12:13PM +0530, Kotresh Hiremath Ravishankar wrote: > Hi Marcus, > > There are no issues with geo-rep and disperse volumes. It works with > disperse volume > being

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

This setup is rather complex, so apologies for the length of this mail. I'm also not at all sure where my problem is, so again I apologise if this is the wrong place to ask. I have a raspberry pi sitting in a remote office (hostname 'pi' below), a VPS hosted on someone's cloud (hostname 'vps' below) and my local laptop (hostname 'laptop'). The remote office then

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial

I am using Apache2.2 + mod_proxy_balancer + mongrel setup and my mongrel.conf file(for apache is like this) 2 3 <Proxy balancer://mongrel_cluster> 4 BalancerMember http://127.0.0.1:9000 5 BalancerMember http://127.0.0.1:9001 6 BalancerMember http://127.0.0.1:9002 7 </Proxy> 8 9 <VirtualHost *> 10 ServerAdmin rags at

I had to setup a new Samba server as a PDC in my job (Samba 3.0) to replace and existing Samba server PDC (Samba 2.2), the old server was running ldap as backend, the new is running tdbsam. I have a lot of bizarre problems when I put both in the same workgroup with other machines. I will give you a list of them: 1) The new server is configured as local master browser and domain master browser,

Hello, I have tested a simple thing in two version of Linux and there was a problem when I have used fwmark as a selector for rule lookups. As you can see below on the Slack, the <test> table doesn''t have any label about fwmark. Is there an iproute problem? -------------------------------------- # cat /etc/fedora-release Fedora Core release 2 (Tettnang) # uname -r 2.6.5-1.358 #

Hi, I am trying to achieve Stateless NAT with ip rule and ip route. Thanks to LARTC doc, I have done it :) But, I have a lot of client wanted access to Internet, setting up 2 rules for each of them is not desirable. For example I have 2 clients: Current setting: [root@son-ag webauth]# ip ru 0: from all lookup local 32760: from 192.168.8.113 lookup main map-to 192.168.250.113 32761:

Here is the situation. There are six different soccer games, developed by Konami, all straight ports from Playstation 2. These are the following: Pro Evolution Soccer 4 (http://appdb.winehq.org/objectManager.php?sClass=application&iId=2750) (PES4) Winning Eleven 8 (http://appdb.winehq.org/objectManager.php?sClass=application&iId=5152) (american version of PES4) Pro Evolution Soccer 5