search for: b2bua

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.ber...

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 Bes...

...ll 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 would be true of any B2BUA. There are two flavors of this problem. In one version, addresses outside of the PBX work fine but calls from different contexts within the same multi-tenant PBX cannot be returned from call history. In the other, all call hist...

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 > everything unde...

-----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 (Darwi...

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

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. |=...

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

...tion = 87 > > Asterisk-Bill-Sec = 82 > > Asterisk-Disposition = "ANSWERED" > > Asterisk-AMA-Flags = "DOCUMENTATION" > > Asterisk-Unique-ID = "1233327083.102" > > Asterisk-User-Field = > "B2BUA-PROVIDER-PRUEBAS" > > User-Name = "SIP/5060-0972ae58" > > Acct-Session-Id = "1233327083.102" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 10.10.149.211 > > > > Fri Ja...

Hello all, It's my try to make some 'emulation' of vovida's b2bua using asterisk. I was in rush while writing it, so I sure there is much code that can be cleaned, great that not too much. :) http://dslmax.boom.ru/asterisk_b2bua_v0.1.zip cdrradius and agi script inside. __ Mike Tkachuk

...> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1">Hi,<br> <br> i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull b2bua server, any suggestion to howto or documentation to asterisk integration and b2b use?<br> <br> ty in advance.<br> </font> <pre class="moz-signature" cols="72">-- Riccardo Cupardo </pre> </body> </html>

Any tips on solving the following performance conundrum: Asterisk 1.8.12.2 running on HP DL360 G5 and G7s tcpdump running to capture UDP 5060/SIP signaling to .pcap files All calls are ultimately B2BUA client -> asterisk -> PSTN Media stays on Asterisk at all times AGI script has exit handler that connects and updates an external database upon BYE from either side. I know that if exit handler hangs around too long, Bad Things (tm) will happen. Oddly, under load (60-100 B2BUA calls), the...

Hi every one, I am using asterisk version 1.6.2...... i did not install mysql data base and when i tried to register a client from SIPp xml file..... it is registered.... My questions are 1. where can i find that registered client? 2. when i type the command "core stop now" it exists and the registered users are not shown why this is happening? 3. Is it compulsary

I understand Asterisk is more like a B2BUA. But when this INFO request is sent to asterisk, asterisk is supposed to bridge the request to the other endpoint, right? In what situation, it decides to send a reply; in what situation, it decides to bridge the request? What is the role of gateway in SIP world, a proxy, a B2BUA or something else...

...;2009-01-30 14:52:50 +0000" Asterisk-Duration = 87 Asterisk-Bill-Sec = 82 Asterisk-Disposition = "ANSWERED" Asterisk-AMA-Flags = "DOCUMENTATION" Asterisk-Unique-ID = "1233327083.102" Asterisk-User-Field = "B2BUA-PROVIDER-PRUEBAS" User-Name = "SIP/5060-0972ae58" Acct-Session-Id = "1233327083.102" NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 10.10.149.211 Fri Jan 30 23:34:08 1998: DEBUG: Handling request with Handler 'Request-...

...you will quickly run into resource problems in medium-sized environments. It also doesn't have a lot of the SIP proxy functionality that SER has. If you're careful, you can configure Asterisk NOT to handle the media stream and still use it for prepaid solutions (using astcc or asterisk-b2bua), and this will save you bandwidth (but you'll still likely run into NAT issues that need to be dealt with somehow) and still let you use Asterisk as an in-between point. Together, Asterisk and SER make a very powerful combination for providing a full suite of services to clientele, and eac...

I am trying to setup asterisk as a registrar and sip server only. Currently When I make calls all my rtp traffic is going through the asterisk server as a B2BUA. Is it possible to turn off this feature and have all my calls RTP traffic going directly to the SIP UA? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/53d0f934/attachment.htm