similar to: why DUNDi ${IPADDR} can not transfer to 127.0.0.1?

---------- Forwarded message ---------- From: ?? <jeffery9@gmail.com> Date: Apr 20, 2006 5:39 PM Subject: why DUNDi ${IPADDR} can not transfer to 127.0.0.1? To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> in the documents, it said that.... ; 'dest' is the destination to supply for reaching that number. The ; following variables

hi, alll,,, there two asterisk box share one realtime database... and all the client is IAX2.. and registery dynamic... there have some question need to confirm.. 1, when i run iax2 show peers,,,there no show the peers that registed with real time... the same as run iax2 show users..there not show any real time users.. 2, if user1 have registed with box1,, how user2 on box1 and user3 on box2

this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150

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

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

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

Hi ! I think that the configure option --with-ipaddr-display doesn't set the IPADDR_IN_DISPLAY define in config.h Here's a small patch to configure.in that should enable the feature (after running autoconf again). -Jarno --- openssh-2.1.1p4-orig/configure.in Sat Jul 15 07:59:14 2000 +++ openssh-2.1.1p4/configure.in Mon Aug 7 08:18:15 2000 @@ -1026,7 +1026,7 @@

Hi, i'm having asterisk with sip working fine, including dundi lookups. The only problem i'm having is that the dundi answer allways contains my internal, private ip. Is there any way to set the targeting ip that is sent out in the dundi answer (to my public ip or any other where i want to receive the call)? Regards, Andreas.

> -----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.

hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+database&diff2=3 says that "Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited",at the same time I find

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 =>

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

I am not sure if I am going to use SIP registration's or just specify the host ip address in sip.conf. Are there any pros or cons to the two? My phones will have a static IP address and won't be changed unless a admin does it. So the logical path would be to just turn off registration on the sip account (in the phone setup). Can anyone forsee a problem to this? Something I will miss

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

I'm looking at finding a way for my Macro(dundi-dundi-test,${ENTEN}) when I dial out on the dundi-test network to return a +101 to my [dundi-test-out] context, if the number being dialed on the dundi-test network does not exist, then I will route the call out using my pstn or voip connection i have. I have a feeling it will have to be the switch => DUNDi/dundi-test that will have to return

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>