similar to: Ser, asterisk and ip2ipgw

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

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

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, When mounting a NFS share and an error occurs for some reason, the NFS share is not unmounted correctly: the local path is used instead of the remote path in the umount_v[23]() call. This patch fixes this. I found this while debugging some problems mounting the root file system on NFS in a test system; I saw this in the logs (/mnt/duplicated is the legitimate remote path): Jun 21

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

The VNC server just listens to the appropriate TCP/IP port and then runs Xvnc which does the actual VNC communication. Ideally I'd be able to do the same thing for RDP then the daemon doesn't get any more complicated, and a bug in the RDP layer can't crash the server. I don't know enough about how NX works but I suspect we could do the same thing as for VNC and RDP. I'm

https://bugzilla.netfilter.org/show_bug.cgi?id=1410 Bug ID: 1410 Summary: STATELESS, rules with notrack into a map Product: nftables Version: unspecified Hardware: x86_64 OS: Debian GNU/Linux Status: NEW Severity: enhancement Priority: P5 Component: nft Assignee: pablo at

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

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

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 >

Hi; First, sorry if this question is mostly netfilter related, than lartc, but I think you guys may have a your opinion about this. I''m using Linux 2.4.x with netfilter packet filtering / NAT on our front-end firewalls (P500 with 1Gb RAM), which are filtering traffic going to our Public Web Sites. The traffic is growing very fast since several months.. The average traffic filtered by

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Dear all, I have been writing during last year at the European Bioinformatics Institute a general unified open source solution for R integration. This work is now available via this link: http://www.ebi.ac.uk/microarray-srv/frontendapp/ The different frameworks and tools of the biocep project are now robust enough for production use. The different APIs are finalized but the documentation is

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

Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia (north africa). I have configure my MFCR2 according argentina R2 settigs : the test call run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2. generally in the begging no errors occur. after

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

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:

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