similar to: Fwd: why DUNDi ${IPADDR} has been transfered to 127.0.0.1?

in the documents, it said that.... ; 'dest' is the destination to supply for reaching that number. The ; following variables can be used in the destination string and will ; be automatically substituted: ; ${NUMBER}: The number being requested ; ${IPADDR}: The IP address to connect to ; ${SECRET}: The current rotating secret key to be used .................... but in my experiences.. i

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 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

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

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

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 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 a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

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

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

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 didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

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 =>

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

>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