similar to: Asterisk B2BUA and Site to Site transfers

Hi Guys, Just need some guidance regarding AD across sites. We have two sites, siteA and siteB. Until about a month ago both sites were running NT4 domains, separate domains but with the same names, let's say thedomain. We classicupgrades siteA to AD and now need to migrate siteB to AD. The sites are connected with a WAN link We think ,the steps involved will be the following -

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

Hi Tom, Thanks for the feedback about my Shorewall evaluation I''ve published a blog today covering general things I''ve observed about the way to combine Shorewall with strongSwan: http://danielpocock.com/practical-linux-vpns-with-strongswan-shorewall-and-openwrt Please let me know if anything is inaccurate or if there is anything substantial that I missed and I''ll

Is there a way to disable the b2bua feature in asterisk. I would like asterisk to work as a sip server and not be involved in the RTP path between phones. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/3716b8e2/attachment-0001.htm

Hi, What are the current best practices for running asterisk as SIP B2BUA? Are there any sample configs online or the books that detail this configuration for the newbies? I'm going to run it behind 1:1 NAT for the clients in the public internet so I will use the externip, localnet, and nat settings. Thanks, Andrew

Hi everyone, any chance * could be used as a b2bua without forcing the media stream through the same box? I would love to do some computing on incoming calls, do things like setting another callerid and the forward the call to another sip UA - all without any audio traversing the * box. Any ideas? Thanks, Thilo

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions and critics welcome! http://b2bua.berlios.de Best regards, Mike

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions welcome! http://b2bua.berlios.de Best regards, Mike

Hi all, I have a strange goal..... the setup: two sites ("a" and "b") both with linux machines running shorewall. a machine at site ''a'' needs to connect to services on a machine at site ''b''. both sites have dsl with dynamicaly assigned ip addresses. site ''b''s ip can be resolved from siteb.dynamic.dns.com (one of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Is the asterisk B2BUA patches useful anymore?? I'm trying to set a prepaid SIP network and the only way seems to get through a patched asterisk with B2BUA functions.. The patches failed, Hunk + problems: I have repaired them, but is it very useful?? Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

Hello. LCR means least cost routing, and it's billing system problem where to route a call, not b2bua's. But currently I dunno any free billing system that support it, so i moved this logic to b2bua. On Sat, 12 Feb 2005 07:05:39 +0330, mohammad <mohammad@mirzaee.net> wrote: > Hi Mike; > Thanks for your new application, but I think it would be better if you put >

Can Asterisk only send and receive SIP packet without media proxy in any time? I am using re-invite but I don't want that the ring back is proxy by asterisk. Someone knows a way to do that? Sebastian

I need to replicate MySQL DB of mulltiple server on SiteA to my DR-Site Site_B... all DB are alocated on RHEL,SuSE,Centos,Debian, FreeBSD servers. I need a script to take Multiple MySql DataBase Backup and then import to SiteB, the replica can be done as cold or hotbackup and cron it Thanks -- Madunix_at_Gmail Sysadmin "Computers are useless. They can only give you answers" - Pablo

Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The

First, background information. We are a large (geographically local) organization with 50 sites, including our HQ. Each site has a Debian Server running Samba in NT-Domain Controller mode. Each site is independant of the next, but are all named <SITE>.example.com. The workstations are connected, and working fine in our sites with the single servers. We had a recent network upgrade that now

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

I''m trying to use my VPS server (single interface of course) as somewhat of a VPN gateway to my other location (which is not accessible directly from some places) where the openvpn server is running, and am kind of lost as to what to try next. I tried a redirect rule, but apparently shorewall didn''t like that (it just failed to start). I tried adding the rules via

HI, I had read from the rsync man page on transferring files using the user's numeric id and gid. I have a user which resides on 2 remote sites, having the same user names, but different uid and gid on both sites. I'm transferring the files as a root user. site A site B ----------------------------------------------- usr_A usr_A uid = 100 uid = 200 file1(usr_A)

Let''s say I have two different geographic sites. They are pretty much identical ie. each site has a machine called web1 which is a web server, etc. Except there are couple site-specific settings ie. outgoing DNS servers are different, SSL certs are different etc. On the puppetmaster I can put in a file called e.g. /etc/sideid which would uniquely identify a site ie. siteX or siteY.