similar to: PJSIP Dial via IP fails

hi! Sorry in advance if I am not on the right mailing-list and sorry if my question is cover somewhere is doc (I didn't find anything). The result from nmblookup (same result on all platform) for a given samba server isn't what I expect. The key part of my smb.conf is: [global] netbios name = ourserver2 netbios aliases = ourserverdev ourserverlab ourserver name

After upgrading to openssh-3.4p1, password authentication is no longer working on my system. I'm running Linux RedHat 6.2 with: kernel 2.2.20 openssh-3.4p1 openssl-0.9.6 pam-0.72-6 pwdb-0.61-0 I've tried it with and without compression, with and without priv sep, and I always get errors like this: Jun 30 19:07:48 sugarfreejazz sshd[1344]: Failed password for randy from 10.10.10.2 port

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hey guys, I'm trying to echo my password into some commands inside of a bash script. But I think I'm going about it incorrectly. Here's the top part of my script: #!/bin/bash pub="~/.ssh/id_rsa.pub" dps_pass="my_pass" ssh="/usr/bin/ssh" scp="/usr/bin/scp" for i in 10.10.10.2{5,6} do echo "xfring key up" echo $dps_pass | $scp $PUB

Hi: First of all, thanks for this great piece of software! I found very clear the presentation sheets used at Linux Expo, Amsterdam. :-) About my problem, I setup everything to connect our two boxes. If I type 'ifconfig' I can see them up. Also the routes. ISIvirtual - Our firewall (iptables) which connects to the Internet through ADSL with a public dynamic IP. (the SSL client)

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

I want to configure a device with three network interfaces where two of them would bridge two segments of the LAN subnet and the third one would be connected to the WAN link. eth0 - 10.10.10.2/24 to be connected to the internet gateway having IP 10.10.10.1/24 (also the default gateway for the device) eth1 and eth2 bridged as br0 with IP address 172.16.100.1 connected to different segments of the

Hi I have recently installed madrake 10.1 on my PC, i wish to connect my laptop running XP to linux machine mainly for internet sharing. Linux machine is connected to internet via ADSL USB modem (D Link 200). I use eciadsl package(driver) for this modem and apparently in the network interface it comes up as tap0 . I have network card installed to my PC which comes up as eth0. This network card

Dear R users, I hav two basic data manipulations questions that I can't resolve. My data is a data frame which look like the following : id type 10002 "7" 10061 "1" 10061 "1" 10061 "4" 10065 "7" 10114 "1" 10114 "1" 10114 "4" 10136 "7" 10136 "2" 10136 "2" First, I

I'm in need of installing two NICs in a machine that are connected to 2 different public networks and I need both IP addresses accessible from the outside. The IPs are part of two completely different subnets. For example: eth0 IP - 10.10.10.2/30 GW - 10.10.10.1 eth1 IP - 20.20.20.6/30 GW - 20.20.20.5 How can I configure 2 gateways on this server? Thanks! -- Matthew

Hello, First of all I would like to apologize if my mail is not clear, this is my first time doing this kind of things ! I'm currently working within a small AD environment within a single forest-domain which is the following : - DC Windows 2016 - Archlinux with samba version 4.12.3 I currently have a standard domain user named 'test' which has the attribute

A few weeks... like in a year and a few weeks: https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/ Some interesting bits in there as well, like: "These rules do not apply to providers that lack control of the network infrastructure necessary to implement STIR/SHAKEN." See also: https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN *Jeff LaCoursiere* STRATUSTALK, INC.

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

Hi, folks I installed samba 3.0.21b-2 with winbind on a Fedora 5 server. I edited 5 files (show below) and join Windows AD by "net join ADS" command. It worked in the first month. I could access to folders with appropriate permission. Then I found I couldn't access to the server by keying-in "\\smbservername". A pop-up Windows box say "Incorrect password or unknown

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hello List; I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled?

I could not find a way to set a max on the calls allowed through a PJSIP endpoint. In case we decide to add it, the we need another reason for the call to fail in the Dial application, something like "limit reached" Am I missing this capability?