similar to: FW: Radius on *

On 12/21/11 11:24 PM, Oksana Tkachuk wrote: > Hello, > Thank you very much for the pointers. > I am are able to create new MDNodes, filled with some constants, and > attach them > to llvm instructions. However, the metadata map is not getting updated > as expected. > For example, instead of the expected new entry If you look at PoolMDPass::runOnModule(), you'll see that

I've now seen this on two different VMs on two different ESXi servers (Xeon based hosts but different hardware otherwise and at different facilities): Everything runs fine for weeks then (seemingly) suddenly/randomly the clock STOPS. In the first case I saw a jump backwards of about 15 minutes (and then a 'freeze' of the clock). The second time just 'time standing

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 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

Hello, Thank you very much for the pointers. I am are able to create new MDNodes, filled with some constants, and attach them to llvm instructions. However, the metadata map is not getting updated as expected. For example, instead of the expected new entry !n = metadata !{some values} we are getting !n = metadata !{null} Do you know what might be wrong? Do we need to enter the MDNodes into the

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Hey Guys, today I upgraded to 4.8-RELEASE-p15. As usual I set IPFIREWALL to default accept in my kernel config file. Config & make weren't complaining so, installed the kernel, reboot and there it was: >IP packet filtering initialized, divert disabled, rule-based forwarding enabled, default to deny, logging disabled Another rebuild didn't work out so... I reviewed

hello i am trying to use res_odbc for sipuser. my connection is working. i have checked using isql. even cdr_odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=>odbc,asterisk,sip_buddies sippeers=>odbc,asterisk,sip_buddies

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. 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

-----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 >

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. |======================> UA2

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

I previously had WPA radius authentication working from my laptop to my home network with the laptop running Fedora Core 6 and the server running freeRadius under CentOS 4.4 (freeradius-1.0.1-3.RHEL4.3). I'm attempting to move my FC 6 boxes to CentOS 5 so I decided to pick on the laptop first. Unfortunately, I neglected to backup /etc before doing the CentOS 5 install (bad Dave, bad

Hi all: I ahve some basic questions regarding the mpd.conf: set radius retries 3 set radius timeout 3 set radius server 192.168.128.101 testing123 1812 1813 set radius me 1.1.1.1 set bundle enable radius-auth radius-fallback Here my radius server is 192.168.128.101 and interanl interface of this mpd server is 192.168.64.65 1) What is this "testing123"? is that key between radius