Sip trunk no sound

Asterisk Asterisk General
1

hello to all

I have a problem when using a sip trunk
to make a call by sip trunk hear the call signal person meets the other side but I did not hear the voice.

I use Asterisk 11.1.2 on centos 6 have the firewall opened with udp / tcp 5060 and udp 10000:20000

Cosola in the asterisk of this result:

  • SIP/441002-00000024 is making progress passing it to SIP/20097-00000023
    • SIP/441002-00000024 is ringing
    • SIP/441002-00000024 answered SIP/20097-00000023
    • Locally bridging SIP/20097-00000023 and SIP/441002-00000024
    • Locally bridging SIP/20097-00000023 and SIP/441002-00000024
      == Spawn extension (default, numberoffphone, 1) exited non-zero on ‘SIP/20097-00000023’

can anyone help

Greetings :confused:

2

This will be a firewall or NAT issue.

Note, please use Asterisk Support for future support requests.

3

hello to all

I use centos 6 with asterisk 11.1.2 I opened the firewall tcp 5060
and rdp 10000:20000!

Can someone help configure iptables nat

regards

4

run this command rtp set debug on and check the media stream address

5

hi ,

open the doors in my firewall tcp 5060 to ip machine and rtp port 10000-20000

bur see when i make comand rdp set debug on
Sent RTP packet to 192.168.0.16:10096 (type 00, seq 051700, ts 012224, len 000160)
Got RTP packet from 89.26.246.158:13808 (type 00, seq 036774, ts 012384, len 000160)
Sent RTP packet to 192.168.0.16:10096 (type 00, seq 051701, ts 012384, len 000160)

regards,

6

RTP ports are UDP not TCP. and the audio is flowing between this 2 addresses

Sent RTP packet to 192.168.0.16:10096 (type 00, seq 051700, ts 012224, len 000160)
Got RTP packet from 89.26.246.158:13808 (type 00, seq 036774, ts 012384, len 000160)

Check your nat setting in your sip.conf file, and also a full sip trace will help you

7

put
externip=your–public–ip in sip.conf
reload and check the calls