PJSIP - No Sound

Asterisk Asterisk SIP
1

I have PJSIP working now, for the most part, as far as connecting and registering goes. The only remaining issue is that there is no sound. I have all of the necessary codecs installed and I tried everything Zoiper’s site suggested, so I believe I need to configure something on Asterisk’s end.

This is the extension I am testing with. I am able to call it and the call plays out, but there is no sound on Zoiper. Is there anything I can try in Asterisk in regards to this issue? I have also tried a few suggestions from similar threads on this forum, but none of those worked for me so far.

exten => 1234,1,Answer()
exten => 1234,n,Playback(conf-adminmenu)
exten => 1234,n,Hangup()

2

No sound can be a network failure. Run tcpdump in front of your asterisk and “rtp set debug on” in asterisk CLI. You should see in both cases IN and OUT RTP traffic.

3

Knowing which ones you have tried is important. I’m also sure most of those threads stress the importance of providing logs.

4

And configuration as well.

5

After doing that, I saw a bunch of logs like this when I made the call:

Sent RTP packet to ****:49881 (type 00, seq 015506, ts 059200, len 000160)

6

Here is my configuration as well:

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=****
priv_key_file=****
local_net=192.168.0.0/16
external_media_address=****
external_signaling_address=****

[1000]
type=aor
max_contacts=1

[2000]
type=aor
max_contacts=1

[1000_identify]
type=identify
endpoint=1000
match=0.0.0.0/0

[2000_identify]
type=identify
endpoint=2000
match=0.0.0.0/0

[auth1000]
type=auth
auth_type=userpass
password=pass
username=1000

[auth2000]
type=auth
auth_type=userpass
password=pass
username=200

[1000]
type=endpoint
context=default
disallow=all
allow=ulaw,alaw,gsm
auth=auth1000
aors=1000
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
transport=transport-tls

[2000]
type=endpoint
context=default
disallow=all
allow=ulaw,alaw,gsm
auth=auth2000
aors=2000
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
transport=transport-tls

7

So you don’t receive RTP packets. Also, not sure that the other end receives those packets, problem is before asterisk

8

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