Hello,
I am trying to use Asterisk 13.15.0 for a faculty project,and I have available a server with a public IP to configure it.I have made 2 basic SIP accounts with 1 extension each.I am using Zoiper Client to test this,and from what I can see (the client and the server cli) the accounts register just fine , the call rings as it should but when i answer it , I am unable to hear anything.
Can you please advise ?
Thank you for your suport
The most likely cause, on the rather limited information provided, is that the RTP is being blocked by a firewall, probably the Linux firewall.
Also note, that in Asterisk terminology you can’t have SIP extensions. Extensions are reserved for the things that appear immediately after “exten =>” in extensions.conf.
If that isn’t the problem, you will need to provide your configuration and the full log contents with sip set debug on, core set verbose 5, and core set debug 5. The full log needs enabling in logger.conf.
Hello david!
First of all thank you for your kind and prompt reply!
I am now going to attach the basic configuration for my 2 SIP accounts.Sorry for the delay , I had a problem regarding server access.
extensions.conf
[default]
exten =>6001,1,Dial(SIP/demo1,120,Tt)
exten => 6002,1,Dial(SIP/demo2,120,Tt)
sip.conf
[default]
[demo1]
type=friend
host=dynamic
secret=parola1
context=default
deny=0.0.0.0/0
permit=0.0.0.0/0
[demo2]
type=friend
host=dynamic
secret=parola1
context=default
deny=0.0.0.0/0
permit=0.0.0.0/0
The Tt ought to inhibit directmedia, although it would be advisable to explicitly select a dtmfmode and/or a DTMF friendly codec.
What is in rtp.conf, and is the firewall open for those ports?
Please also note that, unless both peers are on the same IP address, you should use type=peer, not type=friend. This affects security and signalling issues and won’t cause a no media fault.
Hello,
In the rtp.conf file i have just the default configurations. Regarding the firewall i Just sent an e-amil to the server administrator in order to check the availabilit of the rtp ports.
[general]
rtpstart=10000
rtpend=20000
Another symptom of the issue is, when i call from my phone (via 4G) to an extension, i just have the no voice issue, but if i call from a pc to my phone , i can’t seem to answer the call:i get the accept decline screen i accept, and in a few moments the screen reappears
4G is irrelevant.
The phone may well be aborting the call when it fails to get media in a reasonable amount of time. You would need the logging, to really understand what is happening, and even then you may only see how the call was closed, not why…
Hello,
For those of us with a similar issue I have found the following fix :
I also had the same issue i was using the asterisk server in the VM , the VM was not assigned the address so the RTP Packets were failing to transfer from asterisk to zopier softphones.
I changes the VM network settings from NAT to Bridge mode and i am able to hear the audio