similar to: VoIP with asterisk and x-lite

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan.

Hi all I've only been working with Asterisk for a matter of days but have already grown into a big fan =) Much as I've managed to get internal calling working fine, I have a configuration running on an old PII-233 on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond W6692 PCI Card as /dev/ttyI0. The card works fine in minitel and dials out without a problem.. However try

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Hi all I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr) exten=>401,1,Dial(SIP/phone1,20,tr) 301 is the extension number for phone 2 in asterisk server

--delete option is not working for me. The following is a minimal example. $ mkdir box1 box2 $ touch box1/letter box2/extra $ rsync -a -vv box1/ box2/ sending incremental file list delta-transmission disabled for local transfer or --whole-file letter total: matches=0 hash_hits=0 false_alarms=0 data=0 sent 92 bytes received 34 bytes 252.00 bytes/sec total size is 0 speedup is 0.00 $ tree

Hi. My goal is to use rsync to syncronize box1 and box2. On box2, I have a tapedrive which I write the data I syncronize from box1 every night. On box1, my /etc/rsyncd.conf looks like this: root@box1# cat /etc/rsyncd.conf max connections = 1 syslog facility = local6 [tmp] path = /tmp read only = yes comment = export of /tmp hosts allow = box2 auth users

> Message: 13 > Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST) > From: =?iso-8859-1?q?Eugen=20Cristea?= <tecristea@yahoo.co.nz> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] asterisk to asterisk config > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I would like to set two separate asterisks to talk to > each other. > Any

I have two Asterisk boxes, Box1 has TDM22B and Box2 has a TDM11B. From Box1 I can pick up Zap/1 and dial 300 causing Zap/1 on Box2 to ring. From Box2 I can pick up Zap/1 and dial 100 causing Zap/1 on Box1 to ring. PSTN calls come into Box1. They ring thru to Box2 on Zap/1 via IAX, then if no answer they ring Zap/1 on Box1. When such a call comes in and is answered on Box1, I can transfer to

Hello everybody and Timo. I have the next problem. With LDAP tree like ou=mail | - dc=example1.com,ou=Mail | - mail=box1 at example1.com,dc=example1.com,ou=Mail | - mail=box2 at example1.com,dc=example1.com,ou=Mail ? | - dc=example2.com,ou=Mail | - mail=box1 at example2.com,dc=example2.com,ou=Mail | - mail=box2 at example2.com,dc=example2.com,ou=Mail ? and settings in dovecot

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in

I'm a bit confused how scp works... could someone please explain the local/remote external calls that happen when scp is started... in particular how it relates to ssh on the remote site? To be more specific... I use Kerberos for authentication and I've been working on an ssh wrapper script that checks my Kerberos credentials before running the ssh command. If the credentials are

Hi, I get this message on the latest Debian potato for i386. --> ssh -f localhost xterm sb at localhost's password: --> xterm Xt error: Can't open display: That happens also when connecting to another linux box: --> ssh -f box2 /usr/X11R6/bin/xterm --> /usr/X11R6/bin/xterm Xt error: Can't open display: The $DISPLAY is :0.0 for box1 undefined for box2 after

If a call from one Asterisk box to another comes in, it always goes to the default context because it is treated as IAX guest. I don't understand why ?! Asterisk box 1 iax.conf : [BOX2] type=user auth=rsa inkeys=public_key_box2 context=from-BOX2 [BOX2] type=peer host=dynamic auth=rsa outkey=private_key_box1 Asterisk box 2 iax.conf : register => BOX2:BOX2 at IP_BOX1 [BOX1] type=user

I have 2 linux boxes connected to a switch: box1: eth0 192.168.0.200/255.255.255.0 eth1 public address from ISP dhcp box2: eth0 192.168.0.35/255.255.255.0 box3: eth0 192.168.1.3/255.255.255.0 I want box1 to act as a gateway to the internet (it is doing this now for box2), but also want to communicate from box2 to box3 through box1, and that box3 can use the internet through box1. how can I do