I am stuck in sometime with asterisk encryption.
sip.conf reload without any problem, as dial-plan, registering sip clients - no problem at all
When I call form one zoiper sip account to another wireshark capture tcp eth traffic shows following lines:
0x7ff2bc31e6a0 – Strict RTP learning after remote address set to: 192.168.2.15:12744 0x7ff2bc31e6a0 – Strict RTP qualifying stream type: audio. Strict RTP learning complete - Locking on source address 68.195.13.5:44971 Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘PJSIP/1d’ in macro ‘vm’.
192.168.13.253 - asterisk server
192.168.13.252 - android phone (zoiper)
The problem is no sound on both phones during phone calls. Both phones send packages but not receiving any.
That is the SKYPE protocol involved in it? It suppose to be all line of RTP protocol.
Stefan Crain1,54333 gold badges1515 silver badges2222 bronze badges
kantorkantor
2 Answers
If you are registering through SIP but receiving no audio, then for some reason your higher ports used for RTP are not receiving the data, most likely. Usually these ports are 10000-20000. Make sure both IPs can talk to each other through ports 5060-5061 and the higher ports. Can you display your asterisk CLI output while trying to make a call?
asterisk -vvvvvvvvvvvr
uselessinfoguruuselessinfoguru
Great let's get some details.rtp.conf
No errors reloading sip.This is interesting:
According to the CLI console information everything is in order. Asterisk runs on local IP, no firewall.
kantorkantor