13
votes
Accepted
High availability Asterisk options?
Start by having a look at this voip-info web page: high availability design. It will explain what high availability is and isn't - in the context of Asterisk. (Its easy to confuse high availability ...
- 1,674
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 ...
- 4,766
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 ...
- 68.7k
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 ...
4
votes
Accepted
VOIP T1 vs Fiber
TLDR; If you can get the same SLA with your fiber as you can with the T1, go fiber!
Not all fiber is the same, you can have Dedicated Ethernet, QoS Lines, and Business Class Internet.
Dedicated ...
- 7,816
4
votes
Using the VLAN of a bridged switch
You're mixing layer 2 and layer 3 together and they're two different things. VLAN's "operate" at Layer 2 while subnets "operate" at layer 3. It sounds like you have inter-vlan routing configured and ...
- 109k
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
- 126
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 ...
- 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 ...
- 326
3
votes
Auto DHCP IP Phones to a different subnet
Set up a VLAN
Put your phones onto that VLAN
Put a DHCP Forwarder on the VLAN (your switch should be able to do this). Point the forwarder to the IP address of your existing DHCP server
Put a scope on ...
- 68.7k
3
votes
Accepted
EC2 Server Seems to have port issues
Lack of audio in SIP calls is nearly always due to firewall or NAT issues. All traffic to/from your EC2 instance traverses a NAT, so I'd look into making sure that the requisite NAT setting in Elastix ...
- 109k
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:
[...
- 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> ...
- 6,240
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 (...
- 171
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 [...
- 3,667
3
votes
Register asterisk to sip trunk
First, the register line should have a path set at the end, like:
register => myusername:mypassword:myusername@sip.flowroute.com/84106639
Then do a sip reload or service asterisk restart. After ...
- 141
3
votes
Register asterisk to sip trunk
You are missing two items to be placed directly under your register => entry.
[general]
register => myusername:mypassword@sip.flowroute.com
registertimeout=20
registerattempts=0
As soon as you ...
- 31
3
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 ...
- 336
2
votes
Change vlan settings to Yealink phones at once
I would recomend setting up an autoprovision server. You can use TFTP,FTP,HTTP or HTTPS with v6x and v7x firmware Yealink phones (maybe others too).
High level instruction on how to do it using DHCP ...
2
votes
Make an authenticated SIP call via URI
The format for SIP URIs in general is
sip:user:password@host:port;uri-parameters?headers
So in this case you'd need
sip:myuser:mypass@sip.example.com?to=sip:targetuser%40sip.example.com
Notice the ...
- 45.1k
2
votes
Accepted
Does a SIP trunk relay the RTP stream or just set up the call?
check this line in sip.conf :
canreinvite = no ; allow RTP voice traffic to bypass Asterisk
if it set to yes RTP traffic will _try_ to go directly between SIP endpoints.
If it set to no - all ...
- 894
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 ...
- 312
2
votes
Using the VLAN of a bridged switch
If you need the same VLAN to span both switches, you need to define it on both devices and then connect them using a trunk (which also gives you the ability to carry mutiple VLANs on the same physical ...
- 69.7k
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 ...
- 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 ...
- 197
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 ...
- 197
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 ...
- 3,223
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
- 31
Only top scored, non community-wiki answers of a minimum length are eligible