similar to: Starting From Scratch

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

If you update your source from the CVS, you'll get a new SIP channel that supports a new syntax for SIP calls in extensions.conf If you define a SIP peer in sip conf, like [mysipprovider] ... You can now use dial(SIP/mysipprovider/extension) Where the part "mysipprovider" is related to the sip.conf section. Also, you can dial any SIP URL by

I am using shorewall firewall. My firewall have two interfaces: External interface 192.168.9.254 and Internal Interface 192.168.1.1. I have a printer (HP) 192.168.1.200. I want to share my printer with PCs in subnet 192.168.9.*. I use static nat : /etc/shorewall/nat 192.168.9.250 eth0 192.168.1.200 No No And i open all ports for net to local:192.168.1.200 and all ports for

Exactly the same... ??? [root at plexvm ~]# mount -t cifs //192.168.1.200/mp3 /home/plex/Musique --verbose -o username=plex,password=plex,domain=DERWAEL,ro,auto,vers=3.0 mount.cifs kernel mount options: ip=192.168.1.200,unc=\\192.168.1.200\mp3,vers=3.0,user=plex,domain=DERWAEL,pass=******** mount error(2): No such file or directory Refer to the mount.cifs(8) manual page (e.g. man mount.cifs)

Hi all, I'm tearing my hair off trying to understand the difference between C7 & C8 for mounting a cifs FS with fstab I'm building a Plex media server on C8 and duplicated the fstab entries over from my current C7 installation My data (music & movies) are on CIFS shares on a Synology NAS. The packages cifs-utils samba-client samba-common are installed, up to date etc... These

User & pass are present According to the man pages, workgroup is supported I have changed it to domain, but that didn't change a thing [root at plexvm ~]# cat /etc/fstab # # /etc/fstab # Created by anaconda on Fri Apr 3 14:02:23 2020 # # Accessible filesystems, by reference, are maintained under '/dev/disk/'. # See man pages fstab(5), findfs(8), mount(8) and/or blkid(8) for more

Le ven. 3 avr. 2020 ? 17:54, Jonathan Billings <billings at negate.org> a ?crit : > On Fri, Apr 03, 2020 at 04:00:42PM +0200, Patrick DERWAEL wrote: > > //192.168.1.200/mp3 /home/plex/Musique cifs > > user=plex,pass=plex,workgroup=DERWAEL,ro,auto,vers=3.0 > > //192.168.1.200/videos /home/plex/Vid?os cifs > >

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

That was my initial setup before trying the abbreviations, but anyway: [root at plexvm ~]# nano /etc/fstab [root at plexvm ~]# cat /etc/fstab # # /etc/fstab # Created by anaconda on Fri Apr 3 14:02:23 2020 # # Accessible filesystems, by reference, are maintained under '/dev/disk/'. # See man pages fstab(5), findfs(8), mount(8) and/or blkid(8) for more info. # # After editing this file,

Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to "anonymous" since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan

I was upgrading my debian box and managed to upgrade my dovecot installation at the same time. I wasn't planning on it, but it came along just the same. I took some notes on what I thought I needed to change, but apparently it wasn't enough. I've been running this for quite some time now with pgsql authentication and it's been one of those nice applications that always

hello list, I am attempting to load balance SSL web servers using haproxy on centos 5.7. I am using HA-Proxy version 1.4.18 Here is the stanza in the config regarding SSL: listen https 192.168.1.200:443 mode tcp balance roundrobin option forwardfor except 192.168.1.200 option redispatch maxconn 10000 reqadd X-Forwarded-Proto:\ https

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

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

Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.