similar to: No compatible codecs!

Gouri: close. Try "Ssh-keygen -p -P ''". You might argue that ssh should guess that -P imlplies -p, but that's an issue for your ssh maintainer. Also: you don't ordinarily distribute the private key. You need the PUBLIC key in $HOME/.ssh/authorized_keys on any system you want to access with the private key. Maybe i'm seing your application backward, and you

First, an item to fix: the substitution of "-P" for "-p". All good operating systems are case-sensitive, and many utilities, ssh included, are case sensitive about their options. "-P" is passed along with the "-p" to signal that the next parameter is the passphrase, to enable passphrase setting directly in the commandline. If that's wrong,

Hi, After creating and distributing the private key with "ssh-keygen -P", I am till getting the following error message while I schedule from cron. Only exception is right now , I have only one message which says "You have no controlling tty and no DISPLAY. Cannot read passphrase". Any help is appreciated. Gs You have no controlling tty and no DISPLAY. Cannot read

Hi, I have tried to genertae the key with ssh-keygen -P ( remove the passphrase) And copied it to the traget system. However, it doesn't work. Any insite with the way I am distributing the script. Gs -----Original Message----- From: Martin Pool [mailto:mbp@samba.org] Sent: Tuesday, July 23, 2002 6:53 PM To: Kar, Gouri X. -ND Cc: rsync@lists.samba.org; Johnson, Gary X. -ND; Minyard, Mark X.

Hi, I have tried to generate the key with ssh-keygen -P ( remove the passphrase) and copied it to the target system. However, it doesn't work. Any insite with the way I am distributing the KEYS -----Original Message----- From: Martin Pool [mailto:mbp@samba.org] Sent: Tuesday, July 23, 2002 6:53 PM To: Kar, Gouri X. -ND Cc: rsync@lists.samba.org; Johnson, Gary X. -ND; Minyard, Mark X. -ND

(Gouri: a more descriptive subject line will help you get repsonses in future, and please send your mail to rsync@lists.samba.org. Read <http://www.tuxedo.org/~esr/faqs/smart-questions.html>) On 23 Jul 2002, "Kar, Gouri X. -ND" <Gouri.X.Kar.-ND@disney.com> wrote: > Hi guys, I am trying to schdule a script which makes call to RSYNC over SSH. > The same script works

A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with "make samples", nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same trouble. I have several Uniden UIP200 phones and a single Grandstream BudgetTone

I've got all three CODECs the 300 supports -- G.711u, G.177A, and G.729AB -- enabled, I've changed the order, I've got them all in allow lines in my sip.conf, as follows: disallow=all allow=ulaw allow=alaw allow=G729 From "sip debug" I get the following snippets: =================================================================================== Found description format

hello i want to rout my calls to h.323. i have registered my asterisk with GnuGatekeeper. but it is not routing my call to h.323 channel. he is saying Internal channel initialization failed. Bad binary? can any one check my settings what is problem here thanks in advance kamran exten=>_321XXXX,1,Dial(OH323/${EXTEN}@192.168.0.153:1719,30,r)

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

List, I purchased 2 g729 licenses but I can't get it to answer a g729 call from a cisco router with a vwic card. In the debug output below you will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) when it should support g729 according to the config also listed below. The real odd thing is I can place g729 calls to the router, just not from the router to *. Anyone have any

I'm trying to figure out if this is a nat problem. I have a private network behind a freebsd nat box. The * server is on a static nat, with a private ip of 10.139.10.165. I'm connecting with sjphone as the client from 10.139.10.159. I am calling out using simpletelecom. When connecting directly to simpletelecom using sjphone everything works fine. When I go through * I get

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid="Tobias" <1087006> allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which

Hello, I?m trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn?t ?answered? by my server when I try calling the number that I registered at my SIP provider. I?ve registered with register => John.Doe:MyPass:MyUser@my-sip-provider in sip.conf and if I use ?sip debug? I can see the call is coming in but then nothing

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

Hello All, I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However, I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would

Hello, I'm trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn't "answered" by my server when I try calling the number that I registered at my SIP provider. I've registered with register => John.Doe:MyPass:MyUser@my-sip-provider/1000 in sip.conf and if I use "sip debug" I can see the