similar to: server and 2 uniden phones no ringing

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

Hi...rsync working fine for my fedora 5 box to fedora 9 box. Now i want to take back of my folder /home/rajiv to a windows 98 box...Shared a folder in win98 system for full access but do not know how to take backup...tried with the following but got error message [root@myserver ~]# rsync -aPrv /data/stock/ //192.168.1.75/dir sending incremental file list rsync: mkdir

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

Hi there. I am trying to connect Asterisk to a local danish ip-telephony provider. But is having some difficulties. First I thougt they were related to the provider. But then i started debugging on the Asterisk (aix2 debug) When I make a call using AIX to the provider everything seems to work just fine: *CLI> -- Accepting AUTHENTICATED call from 192.168.1.150, requested format = 1024,

fc2, samba 3.0.9 The samba server is PDC. Temporarily, the server was dhcp, which gave it the IP 192.168.1.64 we then set it to static 192.168.1.150, which is what we want. however, the different windows clients, on and off, on ping and such, resolve INTRANET, the netbios name of the pdc, to 192.168.1.64 still, despite me removing wins.dat. I even tried the following suggestion, to no avail

Ok, I have linux samba pdc wins server 192.168.1.150 behind a cisco router(192.168.1.1). everything else behind that router works fine, clients look at 192.168.1.150 as WINS server. vpn over to netgear router(192.168.2.1), with 192.168.2.x subnet. the netgear tells all dhcp clients to look at 192.168.1.150 as WINS server. clients on 192.168.1.x network can ping clients on 192.168.2.x network

I need help setting up a basic routing between two nic's on my CentOS desktop. I thought I'd understood the basics, but ... I don,t want a nat router, just packets forwarded back and fort. This is what I thought would do the trick, but didn't. Can someone please give me some pointers or links so I can get this right. # echo "1" > /proc/sys/net/ipv4/ip_forward # route

I upgraded from version 2.2.0 to 2.2.2 and since then no one can write to their home directory. Attached is a log (at 3) and smb.conf for reference. Also profiles aren't working for Windows 2000 SP2 machines. Any ideas? Thanks in advance for any help Chris Tooley -------------- next part -------------- # Samba config file created using SWAT # from 192.168.1.50 (192.168.1.50) # Date:

I have a simple DHCP range . option routers 192.168.1.1; subnet 192.168.1.0 netmask 255.255.255.0 { range 192.168.1.60 192.168.1.129; range 192.168.1.150 192.168.1.199; } So I want to add a range 192.168.2.1 -> 192.168.2.254 so I did this: subnet 192.168.1.0 netmask 255.255.254.0 { range 192.168.1.60 192.168.1.129; range 192.168.1.150 192.168.1.199;

Hello, Where can I find a link (or docs) to *working* network config for xen 4.1.2? My tests (s. below) were not successful. Thank you in advance for any hints. Regards, Mark # --- root@xen411dom0:~# cat /etc/network/interfaces # Used by ifup(8) and ifdown(8). See the interfaces(5) manpage or # /usr/share/doc/ifupdown/examples for more information. auto lo iface lo inet loopback auto bond0

Dear rsync devs, I recently concluded a bug hunt to trace why my rsync-ing to an SBC was much slower than the corresponding iperf3-reported speeds. To give a concise summary of the situation, in slow wifi links using SSH with ProxyCommand tremendously speeds up things: $ dd if=/dev/urandom bs=1M count=50 of=sample.data 50+0 records in 50+0 records out 52428800 bytes (52 MB, 50

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

CentOS 4.1/bind-9.2.4-2. I have named serving as a cache DNS server plus SOA for a local intranet zone. The problem I am encountering - over a period of time it stops responding to queries. nmap scan from a different host shows port 53 is visible. I can telnet to the port but all queries to server time out. So much so that "service named status" and "service named

Hi list, We have a strange behavior: a customer Snom300 behind a public FW has contact like contact              : sip:user at x.y.39.147:2048;x-ast-orig-host=169.254.252.1:2048 The phone can place calls but not receive any. Also, qualify give unreachable which seems correct when looking the x-ast-orig-host IP. Problem is that the local IP of this phone is 192.168.1.75 Question: how

I have a Samba share that I've created from a Linux box. I've mapped this sucessfully to drive in Windows 2003 server. I'm also able to read/write to it under Windows explorer just fine. It's using user security so I have to enter the username and password defined in the Linux box when connecting to this share. I then mapped a virtual directory under IIS to the UNC name of this

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br> <br> Using Asterisk 1.4.25.1<br> Using realtime sip_buddies<br> <br> I notice

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101