I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported currently running on kali (pid = 14877) [Jan 22 17:57:00] ERROR[C-00000007]: rtp_engine.c:401 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? [Jan 22 17:57:00] NOTICE[C-00000007]: chan_sip.c:25550 handle_request_invite: Failed to authenticate device ;tag=9a473a54 I opened the menuselect and selected res_rtp_asterisk. When I try to reinstall (recompile!) Asterisk, I do ./configure This says it's ok! But when I put make or make install, I get this error: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" LDFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts make: Entering directory `/etc/asterisk/asterisk/menuselect' make: `makeopts' is up to date. make: Leaving directory `/etc/asterisk/asterisk/menuselect' [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c: In function ‘ice_create’: res_rtp_asterisk.c:2421:4: error: too many arguments to function ‘pj_ice_sess_create’ In file included from /usr/include/pjnath.h:23:0, from res_rtp_asterisk.c:53: /usr/include/pjnath/ice_session.h:736:22: note: declared here make: *** [res_rtp_asterisk.o] Error 1 make: *** [res] Error

I am looking for a (free) VoIP Software that can do some special. I will run this Software at my Linux Computer. The Programm should use my VoIP Account/Number that i become from my ISP. If someone (me) call this number, the Programm give me at my CellPhone after 10 times ring a Menu. In that Menu is Key 1 for run script 1.sh, 2 for run script 2.sh .... Are there any Software that can do this?

How can I manage incoming and outgoing calls directly via a VoIP server with a compatible 5G/4G/3G stick? I want to use a SIM card that offers a flat rate for calls in Europe. The idea is to use it with multiple devices on Wi-Fi and via VPN with a VoIP client. I also want to share the Internet connection via the hotspot function (WLAN/USB). It's okay if the connection drops during a call. I have been looking for a solution for 2 years. Is there any hardware, e.g. an AVM Fritzbox, that redirects mobile calls from a 5G/4G stick to DECT? I haven't found anything suitable so far.

I am looking for a Jingle client for the command line. Currently, I use Jitsi, but my dreams of not needing the GUI for more than just web-browsing are not realized due to my reliance on this program. Does anybody know of a CLI client that supports: - VoIP - Communication with people on XMPP accounts - Text chat not necessary, I have IRC for that. And if you want to make my week: - Full audio controls (Jack, or input/output device selection). The system is Ubuntu 11.04 Server.

I've been toying around with several distros in my endless effort to ditch windows and stick to linux but I haven't been able to work this out. I used to have a desktop where everything just worked but now I have a laptop (acer aspire e 15 e5-575g-52rj) and I'm having a very annoying issue when it comes to VoIP chatting which I regularly use. I'm experiencing this issues with Skype and Discord, everyone who's listening on the same channel or chat says that my mic sounds like it has some sort of crazy amplification but I've checked pavucontrol and alsamixer without finding anything useful. The only thing I saw was that discord uses something called WebRTC and Skype uses Chrome Sound. Dunno if that's relevant or anything. I wanted to know if my mic was the issue so I unplugged it from my computer and tried to use the internal one on linux, it didn't work so I'm thinking it's a driver issue, but I don't really know what to do now. I tried Ubuntu, Fedora, Solus and now I'm on Manjaro and I'm having the exact same issue on every single one of them. Also, I tried listening to my own mic on linux using this command: pactl load-module module-loopback latency_msec=1 and it worked just fine, I mean I didn't hear any of the issues people talked about. I thought it could be discord's or skype's postprocessing software (don't know if that's its name but I didn't know what else to call it) but I would'nt really know how to troubleshoot that. `

I have setup in my router openwrt and I try to perform a call using my provider's VOIP settings that I retrieved for its router that provided. My network settings are the following:

config interface 'loopback'
	option device 'lo'
	option proto 'static'
	option ipaddr '127.0.0.1'
	option netmask '255.0.0.0'

config globals 'globals'
	option ula_prefix 'fdd2:a40d:d919::/48'


config device 'wan_dsl0_dev'
	option name 'dsl0'
	option macaddr XX:XX:XX:XX:XX:XX

config atm-bridge 'atm'
	option vpi '1'
	option vci '32'
	option encaps 'llc'
	option payload 'bridged'
	option nameprefix 'dsl'

config dsl 'dsl'
        option annex 'b'
	option ds_snr_offset '0'
        option line_mode 'vdsl'
        option tone 'auto'        
        option firmware '/lib/firmware/vr9-B-dsl.bin' 

config device
	option name 'br-lan'
	option type 'bridge'
	list ports 'lan1'
	list ports 'lan2'
	list ports 'lan3'
	list ports 'lan4'

config interface 'lan'
	option device 'br-lan'
	option proto 'static'
	option ipaddr '192.168.1.1'
	option netmask '255.255.255.0'
	option ip6assign '60'

config interface 'wan'
    option device 'dsl0.835'
	option proto 'pppoe'
	option username 'username'
	option password 'password'
    option ipv6 'none'
    option keepalive '10'
    option mtu '1492'
	list dns '1.1.1.1'
    option peerdns '0'

config interface voip
	option device 'dsl0.837'
	option proto 'dhcp'

config interface 'wan6'
	option device '@wan'
	option proto 'dhcpv6'

As you can see, voip uses the vlan dsl0.837. Also, I have configured the siproxy like that as well:

config siproxd general
	# Custom options allow using OpenWRT network names, and defaults should
	# work out-of-the-box. If your SIP devices do not REGISTER externally,
	# you may also need to open firewall ports: tcp/udp 5060, udp 7070-7089.

	option interface_inbound lan
#	option interface_outbound dsl0.837

# All other documented siproxd configuration directives are supported. Use
# a UCI 'option' for single-instance directives, and UCI 'list' entries for
# directives that allow multiple instances, per the examples below.

	# Define low-level network devices, overriding interface_in/outbound:
#	option if_inbound eth0
	option if_outbound dsl0.837


	# Enable DEBUG logging for configuration messages:
#	option debug_level 0x00000100
#	option silence_log 0

	# Load two plugins: one that logs SIP call details to syslog, and one
	# that strips out G.729, GSM codecs:
#	list load_plugin 'plugin_logcall.so'
#	list load_plugin 'plugin_codecfilter.so'
#	list plugin_codecfilter_blacklist G729
#	list plugin_codecfilter_blacklist GSM

daemonize 1
masked_host=ngn.hol.net
outbound_proxy_host = ngn.hol.net
outbound_proxy_port = 5060

Then In my computer connected via ethernet to the LAN has ekiga configured like that: But despite successfully connecting to voip, I am unable to perform a call to my cellphone (my provider offers landline call via SIP/VOIP). My provider is Vodafone in Greece. I attempted ta call sip:^my_cellphone_number^@192.168.1.1 (router's ip) but I am unable to do so. So do you know why? My firewall settings are the following as well:

config defaults
	option syn_flood	1
	option input		ACCEPT
	option output		ACCEPT
	option forward		REJECT
# Uncomment this line to disable ipv6 rules
#	option disable_ipv6	1

config zone
	option name		lan
	list   network		'lan'
	option input		ACCEPT
	option output		ACCEPT
	option forward		ACCEPT

config zone
	option name		wan
	list   network		'wan'
	list   network		'wan6'
	option input		REJECT
	option output		ACCEPT
	option forward		REJECT
	option masq		1
	option mtu_fix		1

config forwarding
	option src		lan
	option dest		wan

# We need to accept udp packets on port 68,
# see https://dev.openwrt.org/ticket/4108 
config rule
	option name		Allow-DHCP-Renew
	option src		wan
	option proto		udp
	option dest_port	68
	option target		ACCEPT
	option family		ipv4

# Allow IPv4 ping
config rule
	option name		Allow-Ping
	option src		wan
	option proto		icmp
	option icmp_type	echo-request
	option family		ipv4
	option target		ACCEPT

config rule
	option name		Allow-IGMP
	option src		wan
	option proto		igmp
	option family		ipv4
	option target		ACCEPT

config rule
	option name		Allow-VOIP
	option src		lan
	option proto		udp	
	option src_port		5060
	option family           ipv4            
        option target           ACCEPT

config rule                            
        option name             Allow-VOIP  
        option src              lan       
        option proto            udp               
        option src_port         7070-7080            
        option family           ipv4      
        option target           ACCEPT     

# Allow DHCPv6 replies
# see https://dev.openwrt.org/ticket/10381 
config rule
	option name		Allow-DHCPv6
	option src		wan
	option proto		udp
	option src_ip		fc00::/6
	option dest_ip		fc00::/6
	option dest_port	546
	option family		ipv6
	option target		ACCEPT

config rule
	option name		Allow-MLD
	option src		wan
	option proto		icmp
	option src_ip		fe80::/10
	list icmp_type		'130/0'
	list icmp_type		'131/0'
	list icmp_type		'132/0'
	list icmp_type		'143/0'
	option family		ipv6
	option target		ACCEPT

# Allow essential incoming IPv6 ICMP traffic
config rule
	option name		Allow-ICMPv6-Input
	option src		wan
	option proto	icmp
	list icmp_type		echo-request
	list icmp_type		echo-reply
	list icmp_type		destination-unreachable
	list icmp_type		packet-too-big
	list icmp_type		time-exceeded
	list icmp_type		bad-header
	list icmp_type		unknown-header-type
	list icmp_type		router-solicitation
	list icmp_type		neighbour-solicitation
	list icmp_type		router-advertisement
	list icmp_type		neighbour-advertisement
	option limit		1000/sec
	option family		ipv6
	option target		ACCEPT

# Allow essential forwarded IPv6 ICMP traffic
config rule
	option name		Allow-ICMPv6-Forward
	option src		wan
	option dest		*
	option proto		icmp
	list icmp_type		echo-request
	list icmp_type		echo-reply
	list icmp_type		destination-unreachable
	list icmp_type		packet-too-big
	list icmp_type		time-exceeded
	list icmp_type		bad-header
	list icmp_type		unknown-header-type
	option limit		1000/sec
	option family		ipv6
	option target		ACCEPT

config rule
	option name		Allow-IPSec-ESP
	option src		wan
	option dest		lan
	option proto		esp
	option target		ACCEPT

config rule
	option name		Allow-ISAKMP
	option src		wan
	option dest		lan
	option dest_port	500
	option proto		udp
	option target		ACCEPT

# allow interoperability with traceroute classic
# note that traceroute uses a fixed port range, and depends on getting
# back ICMP Unreachables.  if we're operating in DROP mode, it won't
# work so we explicitly REJECT packets on these ports.
config rule
	option name		Support-UDP-Traceroute
	option src		wan
	option dest_port	33434:33689
	option proto		udp
	option family		ipv4
	option target		REJECT
	option enabled		false

Do I also need to open extra ports as well?

I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m back to asterisk. I installed asterisk on my debian VPS, then installed free PBX. At first i was having issues with getting the freePBX installer to find my asterisk server. After creating an aserisk user on debian and changing the run user-group in /etc/default/asterisk the installer worked. After the FreePBX installation, localhost/ was redirecting to localhost/admin/config.php but was only showing a blank screen. After running fwconsole ma installall the page started working. Despite that, connection to asterisk cannot be established. Running fwconsole start works just fine, but when running fwconsole restart I get get UCP Node Server is not running. When running fwconsole restart again I get Core FastAGI Server is not running & UCP Node Server is not running. **After checking logs for freepbx I found this:**

[2020-Nov-23 14:42:32] [freepbx.INFO]: Connection attmempt to AMI failed [] []

**The UCP error log in asterisk yielded this:**

2020-11-23 13:40 +00:00: { [Error: Can’t connect to MySQL server on ‘::1’ (111 “Connection refused”)] code: 2003 }

2020-11-23 13:40 +00:00: There was an error with MySQL Connection

**ucp_out.log in asterisk displays this:**

2020-11-23 13:40 +00:00: Starting FreePBX…
2020-11-23 13:40 +00:00: { AMPDBUSER: ‘freepbxuser’,
2020-11-23 13:40 +00:00: AMPDBPASS: ‘****’,
2020-11-23 13:40 +00:00: AMPDBHOST: ‘localhost’,
2020-11-23 13:40 +00:00: AMPDBNAME: ‘asterisk’,
2020-11-23 13:40 +00:00: AMPDBENGINE: ‘mysql’,
2020-11-23 13:40 +00:00: datasource: ‘’ }

I am using mariadb, is there a way maybe this user or database does not exist? Thank you for your help! --- Forgot to mention that when accessing Advanced Setting in FreePBX I get a blank page.

The Asterisk wiki says that the color of the CLI prompt can be changed with %Cn[;n]. ie: export ASTERISK_PROMPT="%C31[%H]: " but when I use the export above, I get broken prompt [1;31m[voip]: [1;0m, where voip is my hostname. My terminal is color capable. In fact, I am using colors in zsh and bash. And echo $TERM gives me: xterm-256color Here is a screenshot of my terminal: As can be seen, even the asterisk messages/logs are colored. I have asked on the asterisk mailing list and the syntax export ASTERISK_PROMPT="%C31[%H]: " is correct. Some people reported that color prompt works for them, others had same problem as I am reporting. This leads me to suspect, that the problem might not be in asterisk, but some complex interplay of OS environment, terminal, terminal emulator, etc. I have tried several different terminal emulators: terminator, xterm, gnome-terminal, eterm, konsole. The problem is the same. I use Debian 10 both on the asterisk server and my desktop from which I am connecting. **How can I get colored CLI prompt on Asterisk?**

I've just bought a new VPS with Cent OS 6.6 installed. I'm attempting to install Asterisk 11 on this VPS via command line remotely. I've used the directions [here](http://www.voip-info.org/wiki/view/Asterisk+11+Installation+on+CentOS+6) however I get this error: you do not appear to have the sources for the 2.6.32-042stab102.9 kernel installed when running: cd /usr/src/dahdi-linux-complete* make && make install && make config How can I install this kernel and continue my install?

I am trying to launch the newest skype versions (~)8.57.0.116-r1 (~)8.58.0.93 on my Gentoo and it won't launch, the only error being this line in dmesg

traps: skypeforlinux trap int3 ip:55e99ed962ff sp:7ffc54486f30 error:0 in skypeforlinux[55e99ce7f000+5422000]

I tried chmod 4755 /usr/share/skypeforlinux/chrome-sandbox, but there is no chrome-sandbox in the package's files whatsoever.

When I run the update script for voipbl manually, I get this error from ipset: ***ipset v6.11: Hash is full, cannot add more elements***. I am running it manually because some IPs that are on the blacklist seem to still be getting through the firewall. From the man pages for ipset it would seem to have something to do with increasing the hash size, or the maximum number of elements, but neither of those seem to work for me. Here is the listing for the set: CommandMe-> ipset voipbl -l Name: voipbl Type: hash:ip Header: family inet hashsize 2048 maxelem 200000 Size in memory: 16460 References: 1 Members: This appears to be how many addresses made it into the set: CommandMe-> ipset -l |wc -l 65549 The manual says the default maximal number of elements which can be stored in the set is 65536. I seem to be going over that limit, but cannot get more than 65549 elements in. I've got about 80000 addresses in the blacklist. Am I getting this error because ipset was unable to store the remaining IPs (65550-80000) in the hash? Can someone please point me in the right direction? Thanks!

With Linux Mint 19.1, kernel 4.15, I connect to the Internet using WiFi, and want to share the Ethernet connection with a Grandstream IP Phone . I have done: > click | network connection | wired connection 1 | edit | ipv4 setting > | automatic dhcp | change to | shared to other computer | click | save With kernel 4.4, the connection is OK, but, since changing to kernel 4.8, the phone can make calls, but I don't hear the person on the other side. I don't receive calls, either. I would to know if there is some port to open or something to do with the iptables. Phone Model: Grandstream GXP1625.

I am using Linux Mint 17, and I installed Viber to use as an alternative to Skype. I got the Viber client that is available on their website: https://www.viber.com/products/linux/
Whenever I start a Viber call, my microphone volume automatically gets turned to the maximum, and the person I'm calling tells me that my voice sounds robotic. I can manually lower the microphone volume, but I have to do this every time I start a new call.
This never happened in Skype, so it must be a setting in Viber, but I'm unable to change it in Viber.
I'm thinking there must be some way to prevent applications like Viber from doing this, but I don't know how.
Any help?

I had clone the linphone-desktop project and follow all the steps provided both on the README and in the README.macos from the linphone repository to install all the dependencies, by the way I'm using MacPorts 2.3.4. So, I have an error compiling after I ran the prepare.py script with the --package option. But the thing is that when I ran the prepare.py script without the --package option and compile again it works just fine. Output from iTerm: Install the project... -- Install configuration: "Release" Cannot find source to copy: /Users/pvaldivieso/Downloads/linphone-desktop/OUTPUT/lib/mediastreamer/plugins/*.*.dylib error: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/install_name_tool: can't open file: /Users/pvaldivieso/Downloads/linphone-desktop/WORK/PACKAGE/Linphone.app/Contents/MacOS/Linphone-bin (No such file or directory) /usr/bin/patch: **** Can't find file /Users/pvaldivieso/Downloads/linphone-desktop/WORK/PACKAGE/Linphone.app/Contents/Resources/share/themes/Quartz/gtk-2.0/gtkrc : No such file or directory pkgbuild: error: Component path "/Users/pvaldivieso/Downloads/linphone-desktop/WORK/PACKAGE/Linphone.app" does not exist. [100%] Completed 'TARGET_linphone_package' [100%] Built target TARGET_linphone_package It goes all the way to the end and fails. I think it's related to mediastreamer and not finding some .dylib but I'm not sure. My question is: how to generate the installation package for Mac OS X?

Now that Skype got discontinued for Linux and [only the 5 beta version is avalable](https://blogs.skype.com/news/2017/03/01/the-skype-for-linux-beta-version-5-0-is-now-available-for-download/) , Linux users have a problem, which is how to communicate with Mac and Windows users. There have been lately a lot of questions regarding the installation of Skype and alternatives for Skype but I think the real problem comes when you 1. Absolutely need to use Skype because everyone else you work with uses Skype 2. Need video for the conference calls with those people 3. Need to share your screen while on the conference call As far as I know, you can't do numbers 2 and 3 if you have to follow 1. That is, there is no way you can natively use Skype to participate on conference calls held from Windows/Mac computers since Skypeforlinux [lacks support for conference calls and outgoing screen sharing](https://support.skype.com/en/faq/FA34656/more-information-about-skype-for-linux-beta) . (There are even problems for [regular video calls!](https://superuser.com/questions/1222703/is-video-call-available-on-skype-for-linux-beta-or-web-skype-com-with-linux-chro)) My question is, if you absolutely have to use Skype on Linux, is there any way around this? The only answer I could come up with was Skype+wine, but it doesn't appear to work. I have spent several hours lately trying to [install Skype 7 using wine](https://appdb.winehq.org/objectManager.php?sClass=application&iId=1592) but the installation fails with "connection problems". Installing [Skype 6](https://askubuntu.com/a/722140/375797) also doesn't work because the link seems to be broken. Is there any way around this that I'm missing? __EDIT__ As of today version 5.4 Beta is releases which apparently fixes the conference call issue. Although the screen sharing is still not possible. __EDIT 2__ Seems like skype preview now has all functionalities, making my question obsolete.

I need to set up an asterisk voicemail inbox that: * Works normally during business hours. * After business hours, calls to this line go straight to voicemail after a special "we're not in right now" recording plays. I think it should be possible to set the recording using custom time zones, but I'm not sure that is the best way to do it. I also don't know how to control "rings before voicemail" based on a schedule. Is this possible?

Criteria: - Makes audio/video calls - Encrypts the whole traffic (using good encryption) - Is cross-platform (including Windows 7, etc.) - Runs on modern Linux distributions (Fedora, Ubuntu, etc.) - Runs on OpenBSD Does anybody know a good Free and Open-Source alternative to Skype?

I'm trying to configure OpenSIPS with OverSIP so that I could make a SIP call between a webrowser and a sip client app like Yate or Linphone. The call between two client apps (Yate/Linphone) works perfecly and between two browser clients (sipml5) also works perfectly. But when I try to make a call from sipml5 to Yate/Linphone than it fails and when I call from Yate/Linphone to sipml5 than it rings but terminates the call when I allow the browser to use my audio/video devices or I try to answer the call. I might be wrong but I think its a missconfiguration in opensips or oversip. What can I do to make this work?

On this site I clicked on the support link and a phone number popped up. It was formatted via the tel: protocol, and as such was underlined and highlighted like a web link. On my phone, clicking on such a link opens my default dialer and places the call. When links are not formatted via this protocol, my browser's Google voice plugin usually auto-detects the phone number and gives me a chance to call via Google voice. It would be nice if I could also do this for tel: formatted numbers, or better yet, as I don't always have a voice session open, set tel: formatted numbers be handled by yate. **How can I fix my browser's handling of the tel: protocol links such that it opens in yate or google voice?** - In Chromium I get a warning saying that xdg-open will be used to handle the link, but when I click 'Launch Application' nothing happens. - In Firefox, clicking on the link leads me to a blank page with the url tel:18003744432 - In Konqueror clicking on the link produces this error: Access by untrusted page to tel:18003744432 denied. I'm using Linux Mint 16 KDE x64.

is this the typical Asterisk topology ? On my home system, I have the Asterisk server, tleilax, and the client, doge, connecting into my router -- so that they're both on the same network. The IP addresses are: router 192.168.1.1 tleilax 192.168.1.2 doge 192.168.1.x (usually 3) Based on the above image, I don't quite understand what the topology should look like. What if you have multiple clients? They can't all connect to tleilax unless you put in a ton of NIC's! Here you see multiple clients connected to an Asterisk box: but no mention of a switch... Pardon, if it's in the definitive guide , I missed it. (I have the 4th ed.) This diagram: well useful, doesn't help me to understand the network topology of a simple setup, but with multiple clients..