similar to: Call history problems from B2BUA

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

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

-----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. 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 >

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

I don't think "very carefully" was the answer that the original poster was looking for. I would suggest one of the multi-tenant GUIs out there, or map out your call flow in a diagram, convert your diagram into the dialplan, then test, test, test. Make sure to test everything, including invalid extension, #, timeout, etc... Another strategy is to use a GUI on a dev box and then

Hi all, We want to offer hosted PBX services to some of our clients (maybe 10-20) and were wondering if it makes sense to get a software package capable of handling multiple virtual tenants or if we should just create multiple virtual machines in our server each running a single- tenant license of the software. We have been researching virtual PBX software for asterisk for a couple of

A friend of mine runs a small office building, 10-15 tenants. Each have their own company, their own thing, renting space from him. His main PBX is getting dated and his tenants are complaining. I was telling him about Asterisk but his main concern is he doesn't want to have to always be the one to add/remove extensions, or change the IVR hours or whatever. Does anybody know of a free or

Hello List - I've been slowing growing my extensions.conf file and have been wondering how everyone manages their systems. I currently have my main extensions.conf where I reference my sub extensions (for tenants or customers) files using the include statements and define my global variables. Today while watching the asterisk console I noticed a call from a voicemail user bounced into

Hi, I''ve dug into some previous discussion on this list about having multiple clients running on a single Rails site. I can see how to work this out with a single project type. For example, my tenants all have their own CMS in a single Rails environment. What if 3 of my clients have a CMS site, 5 have an e-book site and 2 a dating site. Is there a way to house these websites within a

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

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

Hi All, I am seeking input from anyone who may have seen a similar configuration and dealt with similar issues to what I'm experiencing. Configuration: - 2 sites (site A and B) - Asterisk 1.2.23 on each site (Trixbox) - Internet 512/512 symmetric at each site, dedicated to VOIP calls only. - IAX trunk between the sites, with data travelling across the 512/512 Symmetric link - PSTN

Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek -

Dear ALL: I find a program named "asterisk_b2bua" on http://developer.berlios.de/projects/b2bua/ And I also download them(two components) and try to test it. But I have not enough knowledge about asterisk. It seems a Software PBX. Does asterisk_b2bua work? Does anybody ever try it? I have questions about my scenario. |======================> UA2

Thank you, Louis, for your reply. By simply asking me to provide outputs of the aforementioned files, I found the cause of my first problem (auth failing). It was my /etc/hosts file on dc1. All of them should look like this, and indeed DC2 and DC3's *did* look like this: # cat /etc/hosts > 127.0.0.1 ? ? ? localhost.samdom.mycompany.net ?localhost > 192.168.3.201