similar to: voice mail - sip:notify message

Hi, all, I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below: Softphone1<-------------->Asterisk SIP<------------>Softphone2 (User Agent) (Proxy) (User Agent) 155.69.xx.xx 155.69.yy.yy 155.69.zz.zz zhou mysipproxy.com

Hello, I'm using the following Dial command syntax: Dial*(SIP/peer/exten!sip:xyz at xyz.com <sip%3Axyz at xyz.com>*), the SIP URI after the '!' mark should be set as To-URI in outgoing INVITE from Asterisk. It works, but problem is that To-URI formatting is a bit messed up, It looks as follows: *sip:sip:xyz at xyz.com <sip%3Asip%3Axyz at xyz.com>*, it seems that Asterisk

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Hello I want to unlist the attached element getting only the first element in each element of the list. The last element of the list looks as this: [[5065]] [[5065]]$Pluv3Meses [1] 274.4 [[5065]]$PluvMesesMedio [1] 378.2667 [[5065]]$Pluv2UltimosMeses [1] 23.33333 So I would like to get for each element of the list the element called Pluv3Meses. The whole list has 5065 elements but when I try to

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 ->

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

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 > >

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

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

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, 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:

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