[asterisk-users] DUNDi problem: offline peers still in request EID/EID_DIRECT field?
hi all! I have the following situation: 1 2 ¦¦ ¦¦ 3--4 ¦¦ ¦¦ 5--6 where 1 ... 6 are nodes and every direct neighbor is specified as a dundi peer (in *). When I start a dundi request, every queried node is mentioned in the dpdiscover. For example 1 sends a discover to 2 and 3, so 2 sees in the EID or EID_DIRECT field that a discover has also been sent to 3. So much for that. Is this field also filled with the neighbour peers even if they are unreachable/offline? 1 ---x 2 ¦¦ ¦¦ 3x-4 ¦¦ ¦¦ 5--6 My problem: When links break (e.g. 1-2 and 3-4) I have the problem that 6 doesn't forward the query (received from 5) to 4, because 4 is mentioned in the EID or EID_DIRECT field even though it is not possible that this peer could have been reached. Is this a problem of the protocol or can I fix this by setting a special option in *? Thanks for helping. Best regards Andre___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DUNDi weight support in * - HELP!
Hi all On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? Thanks! Andre ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DUNDi weight
Hi all On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? Thanks! Andre___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX-configuration
Hi all I have a network with nodes with different network-interfaces (e.g. node17 with interfaces A and B and node18). Asterisk listens to 17.A, 18's DUNDi knows 17 by knowing ip B. When I start a DUNDi request from 18 to 17 I get a response from A via B. So B knows that the number can be reached at 17.A which is correct. A --- B -- --+ 17 +-+ 18| ----- But when I call this number and debug the IAX-transmissions I get something like: NEW from 18 to 17.A AUTHREQ from 17.B to 18 (f) INVAL from 18 Why comes this AUTHREQ from 17.B and not from 17.A? Why can't I directly speak with the main IP over IAX? The whole thing works great when DUNDI is connected directily to the IP ASTERISK listens to. What can I do? Thanks for your answer. Best regards André ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Ethernet Interfaces
Hi folks I'm emulating a network of several nodes running asterisk. All nodes have multiple ethernet interfaces and are directly connected to their neighbours (ping works). I can lookup also the subscribed phones with DUNDi from neighbour or remote nodes. But when I want to dial the looked up phones some connections don't seem to work any more. There seems to be no pattern, some nodes can communicate over multiple ethernet interfaces, others over none. Can someone imagine what happens here? Thanks André___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help on asterisk sipp
Hello! If ou mean SIPp, the testing tool for the SIP protocol or kind of a call generator for Asterisk PBX, have a look at http://sipp.sourceforge.net/doc/reference.html cheers - Original Message - From: khawla khawla To: asterisk-users@lists.digium.com Sent: Monday, May 28, 2007 3:09 PM Subject: [asterisk-users] help on asterisk sipp Good morning I was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it? If you can explain to me the principle, I would be extremely grateful. Thank you very much in advance. -- Lancez des recherches en toute sécurité depuis n'importe quelle page Web. Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! Essayez-le maintenant ! -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call interruption
Hello all Could someone tell me what happens with running calls when reloading the whole asterisk config files? I think SIP-calls are not interrupted because of the protocol architecture (signalling vs. media) but what's with other kind of calls like h323 or over analogue interfaces? are they interrupted? I'm quite new with asterisk, so excuse this probably trivial question... Andre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
