similar to: DUNDI Sip authentication failure

Hi All, I've been banging my head on a small dundi problem... I have two * servers setup, both have almost identical dundi.conf files: root@tsjonge:/opt/asterisk/etc# cat dundi.conf [general] department=thuis organization=pipsworld locality=Amsterdam stateprov=NH country=NL email=remco@pipsworld.nl phone=+31207508308 ;bindaddr=0.0.0.0 ;port=4520 entity=00:02:b3:49:69:5e ttl=16

>From the reading and testing I have done it doesn't look like SIP supports a username and password in the Dial string. I currently have the following mapping. priv => dundi-extens,0,SIP, dundi:pass at 1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial On the sending side I see NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi' and not

Well, I've been dicking around with DUNDi for about 4 hours now. I have two systems that I am trying to get to peer with each other. Queries are working one way, but not the other. The server, pbx1, that is refusing to deliver any queries and logs what's below to the console. Why??? I've checked all the config files more times than I can remember. The 'RFC' for DUNDi at

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

I have three Asterisk boxes. Each has the following in dundi.conf: 180net => dundi_local,0,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx1,1,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx2,2,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx3,3,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial My iax.conf on all three

Hello, I have two asterisk servers and I would like to merge their dialplans. I thought DUNDi would be a natural choice. I created the following configuration on the first server: iax.conf Code: [dundi] type=user dbsecret=dundi/secret context=dundi-local dundi.conf [general] ttl=4 autokill=yes cachetime=30 entityid=00:06:5B:8E:B0:08 secretpath=dundi bindaddr=XXX.XXX.XXX.XXX port=4520

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

Hi few weeks ago I read about redundancy (HA) of asterisk boxes using DNS, DUNDi, so I decided to give it a try. OS FreeBSD 6.1-RELEASE, asterisk 1.2.7.1 on one peer I get: lk110*CLI> dundi show peers EID Host Model AvgTime Status 00:11:43:3d:69:e6 195.28.109.37 (S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0

[priv] type=3Dfriend dbsecret=3Ddundi/secret context=3Dlongdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > >

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Thursday, June 15, 2006 10:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] DUNDi Not Able to > HandleComplexFailoverSituations > > > Is it possible for you to explain in more detail the > situation involved.

Can any one help? Thanks. we have two * servers (Version 1.2.1) and one 1.09 server. Calls between these two 1.2.1 servers have odd behavior. But call from 1.09 to 1.21 server working fine in either situations. See below pls: Local server iax.conf [tosyd] username=mel type=peer secret=xxxx host=198.168.2.66 remote server iax.conf [mel] type=user secret=xxxx host=198.168.2.67

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a "bye". Bellow is the log of the internal call: --

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000