similar to: load balancing

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

Hi All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the

Hi All, if i do this setup: |---[ext 100] |--[router/nat gw]--| | |---[ext 101] | [asterisk]--[internet]---| | | |---[ext 200] |--[router/nat gw]--|

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

Hello, Here is how I do this. The prerequisits are: - MySQL to hold the extensions realtime database. MySQL is synchronized among all servers using the Master/slave replication model. - The phones are spread by some external algorithm over the Asterisk servers (statefull load balancer, statically defined in the config file of the phone, etc.). The idea is to locate on which server the

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

WINE 0917 cannot install InstMsiA [lando@localhost WINE_0917]$ wine /home/lando/WINE_0917/InstMsiA.exe fixme:msiexec:main /regserver not implemented yet, ignoring fixme:msiexec:main /unregserver not implemented yet, ignoring

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Hi all, From the web, I can find a table scheme of sipusers for ARA using. However, I can't find any meaning of each field, especially for the field regserver which is new in the table. Can any tell me more detail about the usage of each field? CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `host` varchar(31) NOT NULL

Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay.

Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

Hi If we have a shared RealTime database for sip registration of multiple Asterisk servers, is there a way to realize which Asterisk server registered sip phones ? RegardsM.Shirazi? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150817/62c5cd3c/attachment.html>

Hi list gurus, long story short we have firewall machine which is the default gateway for our clients and firewall send traffic out to Internet via cisco router. On cisco we have two serial interfaces 1Mb and 2Mb. On firewall #route add default gw xxx.xxx.xx.xxx (for 2mb) #route add default gw xxx.xxx.xx.xxx (for 1mb) and the same rule for Imb link route packets via these two links. However I

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on