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10 votes

Why do we even need a PBX software like Asterisk?

The PBX is going to handle all of your endpoints, your hunt groups, your schedules, your auto-attendant, call flows/trees, recordings, etc.. Your SIP provider gives you phone lines. The PBX allows you ...
Safado's user avatar
  • 4,786
8 votes
Accepted

Add unmanaged switch to VLAN port

If you are using purely port-based VLANs, then what you have described here will work. Port-based VLANs do not need vlan-aware equipment attached to them. However as soon as you start to add "Voice ...
Mark Henderson's user avatar
6 votes

Freeswitch drops calls after 32 seconds

So I solved this. I captured packets for calls that were working and calls that were not. It appeared Freeswitch was actually asking the clients to respond on an IP address that was similar to my ...
MikeKulls's user avatar
  • 499
5 votes

Asterisk ghost calls

It's a brute force, asterisk servers always get that kinda thing if connected to the public IP. My solutions are : Install fail2ban, fail2ban will set the iptables and reject ip with continous fail ...
Delly Fathurachman's user avatar
4 votes
Accepted

Zoiper softphone cannot register to Asterisk with IAX2: cause connection refused, cause code 29

Try adding the valuehost=dynamic to your guest config in iax.conf
Marco Zink's user avatar
4 votes

Why do we even need a PBX software like Asterisk?

If you just want a simple phone, a VoIP equivalent of just renting a single phone line and plugging a dumb phone into it, you don't need the PBX, just a Sip Trunk. If you want the features that the ...
ccomley's user avatar
  • 76
4 votes
Accepted

How to screengrab from Cisco SPA504g phone?

I'd given up on this, but stumbled across the answer: http://*phone-ip-address*/admin/screendump.bmp Ref from Cisco forums. Tested today on SPA504G with software version 7.6.2a. It's correct right ...
jonathanjo's user avatar
3 votes

Asterisk ghost calls

This happens due to automated scanners, which probably are trying to bruteforce your passwords. To get rid of such calls, you should disable anonymous users by placing the following options into the [...
Anubioz's user avatar
  • 3,725
3 votes

Why do we even need a PBX software like Asterisk?

You absolutely can do this. Whether it's appropriate for your situation is a different matter. Essentially, SIP trunks are like POTS lines or T1/PRI trunks. They are just dial tone. They can be used (...
Josh Richards's user avatar
3 votes

Call from '201' to extension '202' rejected because extension not found in context 'test'

Right now, you only have 1 extension defined in your [test] context ... Extension 20. If you want to be able to place calls between 200 and 201, you'd need to have something like the following: [...
MichelV69's user avatar
  • 249
3 votes
Accepted

Logs concerning failed IAX2 registration attempt missing on Asterisk server

There are multiple logs in /var/log/asterisk, check them all Debugging from the console You can also start asterisk from the console and enable debugging for iax2 # asterisk -r asterisk*CLI> ...
Ryan Babchishin's user avatar
3 votes

Register asterisk to sip trunk

First, the register line should have a path set at the end, like: register => myusername:mypassword:[email protected]/84106639 Then do a sip reload or service asterisk restart. After ...
viktike's user avatar
  • 151
3 votes

Register asterisk to sip trunk

You are missing two items to be placed directly under your register => entry. [general] register => myusername:[email protected] registertimeout=20 registerattempts=0 As soon as you ...
alta3's user avatar
  • 31
2 votes

FAX on VoIP line does not work

Short answer: Use G.711 Codec and turn off T.38 support in your Router. If that still fails, try lower baud rates in your Fax machine. In the case of German Telekom, T.38 isn't supported. Your router ...
NoMad's user avatar
  • 312
2 votes
Accepted

Test if SIP ALG is disabled on router

There's no quick test - you would need to be able to examine the SIP packets after they've been sent by the router to the host system, and see if the payload has been tinkered with. Ask your SIP ...
ccomley's user avatar
  • 76
2 votes

Polycom SoundPoint IP 4000 won't power on. How to isolate power supply?

A Polycom SoundPoint IP 4000 SIP can be powered using ether power-adapter or a standard POE connection. The VVX 601 phone should use the same power supply as the 4000 phone, as most more recent phones ...
RunThor's user avatar
  • 209
2 votes

Asterisk ghost calls

As you stated this is an issue with home/remote users primarily it is most likely they you would see two things. 1- This SIP invite they receives has nothing to do with you Asterisk server. I've seen ...
RunThor's user avatar
  • 209
2 votes
Accepted

ASA QoS for VoIP traffic

Usually, you want to police the priority queue (VoIP). Give it a guaranteed minimum bandwidth, and police everything above that bandwidth. Not policing the priority queue lets the priority traffic end ...
Ron Maupin's user avatar
  • 3,273
2 votes
Accepted

Mikrotik SIP trunk routing

I just had to specify my DSTNAT rules for each ethernet port, instead of having using generic incoming rule
Johan Marais's user avatar
2 votes
Accepted

Where to download Microsoft Lync Server 2010 or 2013

Short answer: Lync 2010/2013 are replaced by Skype for Business. Therefore you couldn´t any longer find them. But as Skype for Business also offered to manage the firmware for Lync Phones you should ...
BastianW's user avatar
  • 2,888
2 votes

Do I need STUN if only one party is behind NAT?

NAT devices change the address and ports of IP packets. However SIP messages contain ip address inside the payload of the IP packets that are not managed by NAT devices. SIP headers as Contact or Via ...
J.M. Robles's user avatar
  • 1,090
2 votes
Accepted

Causes of RTP jitter at the server

Finally figured this out! TLDR: disable power management on the host. Despite the low CPU usage, we still figured this was something to do with CPU load. So, we were experimenting with loading down ...
miken32's user avatar
  • 964
2 votes
Accepted

How to enable asterisk Call Completion on Busy Subscriber (CCBS)?

To have Asterisk enabling CCBS after a CallCompletionRequest the busy device need to answer the SIP INVITE with a 486 Busy Here response. That is, Asterisk have to dial the extension even if it have ...
Geovano LQ's user avatar
2 votes
Accepted

Anybody familiar with the Shoretel database?

The database you search for is named shorewarecdr (from CDR - Call Detail Record). It should have tables call, connect and few other. Check there.
Romeo Ninov's user avatar
  • 5,957
1 vote

How to reset a Cisco ATA 186 to factory defaults?

The ATA must be powered on and disconnected from network. Connect an analog phone with full keypad (Unhooked) Press the function button on the top of the ATA Enter the following: 322873738#*​ Enter * ...
Overmind's user avatar
  • 3,171
1 vote
Accepted

Unable to dial SIP phone connected to 3CX system via Asterisk

Make sure to add the fromdomain=<your 3x domain> and qualify=yes to your sip.conf, and it's type=peer for your task. Final sip.conf register=>400:[email protected]/from-3cx [3cx] type=...
Anubioz's user avatar
  • 3,725
1 vote
Accepted

Connect Asterisk system to SIP provider with Asterisk dynamic realtime

The comments in the default version of /etc/asterisk/sip.conf pointed to a solution. An entry of the form register => user:[email protected] is apparently equivalent to a [device] ...
Drux's user avatar
  • 706
1 vote
Accepted

Check Asterisk dynamic realtime (ARA) configuration with CLI

your answer is here https://stackoverflow.com/questions/24166994/no-show-peers-after-configure-realtime-asterisk-with-odbc Mainly it is about the rtpcachefriends option which must be set right.
Aviator's user avatar
  • 126

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