similar to: Asterisk B2BUA patch useful??

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 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

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

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

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

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

Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK,

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

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

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list. I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm using two TC400B cards for transcoding. When I reach nearly 100 simmultaneous calls, the CDR radius packets are being duplicated and I'm getting this message in the asterisk console : cdr_radius.c:227 radius_log: Failed to record Radius CDR record! I'm also using the radiusclient-ng 0.5.6

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Sorry, but why u r using the Radius with the CDR? Not enough to access the CDR in the /var/log/asterisk/cdr-csv/Master.csv? Also, what kind of Radius u r using? Any suggested link? Regards Bilal > > Hello list. > > I'm having some problems with the CDR Radius in my > Asterisk 1.4. I'm > using two TC400B cards for transcoding. When I reach > nearly 100 >

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? Thank you, Wei Date: Fri, 18 Mar 2005 12:51:28 -0600

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,