similar to: wrong password on authentication for INVITE

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

I've got a problem with a SIP Account I am trying to dial in with. The correct extension rings but when I pick up the call is not made and I get a busy signal. Dialing out works just fine - just calling this number doesn't seem to work. Any pointers? Thx Michael excerpt from sip.conf: [general] context=default port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0

In an attempt to connect our Asterisk 1.6 phone system with another phone system called "Broadsmart", they gave me credentials to register to. Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365) watermelon*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time {broadsmart_ip}:5060 N

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

I was just looking for an easy way to convert between COM datetime and chron datetime (both ways.) I found examples on the list, but they involved origin. Does anyone have functions for converting COM datetime <-> chron datetimethat work "safely"? David L. Reiner > -----Original Message----- > From: Gabor Grothendieck [mailto:ggrothendieck@gmail.com] >

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

I'm sorry to waste bandwidth. I re-read the console message for the tenth time and finally noticed R was looking in rw2001pat for libraries. Looking at my Env vars I see I set R_LIBS to look there. Changed R_LIBS, fixed problem. DOH! So now if I want to use several versions of R simultaneously, what do I do. I set R_LIBS so it would look also in C:/R/extra for some added packages. Thanks and

Hi guys, sorry for the long e-mail, i'm only trying to give as much information as i think is relevant to my problem (console log, sip.conf and extension.conf parts). i've been practicing with callback for a while, but i'm at a dead end. I hope somebody can help me to move on. i have troubles getting two calls bridged together. Scenario is the following: - asterisk calls my

Could someone please tell me what I did wrong to create this message or what I should do to correct this problem? I downloaded 2.1.0 Windows binary and installed into C:/R/rw2010, using the installer. I ran md5check.exe in C:/R/rw2010/bin/ and got "No errors." The problem is this: When I start up Rgui.exe from its shortcut (target= C:\R\rw2010\bin\Rgui.exe --save -sdi, Start in

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

But that doesn't put the result into the history buffer, to be written to a file only later when I savehistory(filename). Bert Gunter also suggested ?capture.output and ?textConnection, but I cannot see how to get text into the history buffer as comments, but with evaluated expressions (values). I know how to use paste, sink, write, etc. but nothing that I can see inserts into the history

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Hi David, I tried the following and get the below error messages.... con = blpConnect(show.days="trading",na.action="previous.days",periodicity="da ily")# connecting Bloomberg > dat <- blpGetData(con,"US4009703799 Equity","PX_LAST",start=as.chron(as.Date("01/01/2005",

I have a call from a call file: Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 20000 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On