I’m fairly new to Asterisk and have a problem regarding audio. I have recently set up a VPS with CentOS 7 and Asterisk 16.9.0. It’s a basic, fresh install I did by compiling Asterisk myself. I used the instructions from the book “Asterisk: The Definitive Guide (Fifth Edition)” to setup everything. Endpoints are setup using a realtime connection with a MariaDB database that’s on the same VPS. The VPS only has one network interface with an external IP.
I was able to make endpoints and register them using several different softphone clients on different hardware devices. The only issue is, I get no audio when calling the different endpoints. I can see that the dialplan I made is working properly (to my knowledge). When calling the endpoints, they ring and I can pick up. I have tested this on several networks and clients, but I still get no audio. Also made an extension that plays the Hello-World sound, but sadly I get no audio here either. Here is some extra information:
Output of the firewall to see which ports have been opened:
sudo firewall-cmd --list-all public (active) target: default icmp-block-inversion: no interfaces: eth0 sources: services: dhcpv6-client http https sip sips ssh ports: 5060/udp 10000-20000/udp 5060/tcp 4569/udp 5061/udp 5061/tcp protocols: masquerade: no forward-ports: source-ports: icmp-blocks: rich rules:
I contacted my VPS provider to ask if certain traffic was blocked by default, this was not the case. Also tried disabling my firewall temporarily, did not work. Also put SELinux on Permissive temporarily, also no results.
Here is the output of my pjsip.conf.
[transport-udp] type=transport protocol=udp bind=0.0.0.0 [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/home/asterisk/certs/self-signed.crt priv_key_file=/home/asterisk/certs/self-signed.key
In this case I’m not using TLS yet to make the call. The endpoints are both on the transport-udp protocol.
Here is an example of the live logging when I make a call to the Hello-World dialplan:
== Setting global variable 'SIPDOMAIN' to 'sip.example.com' [Mar 26 10:40:11] WARNING: res_format_attr_siren7.c:52 siren7_parse_sdp_fmtp: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring. -- Executing [200@sets:1] Answer("PJSIP/0000f30B0B02-00000004", "") in new stack > 0x7f6484017520 -- Strict RTP learning after remote address set to: <INTERNAL IP CLIENT>:4000 [Mar 26 10:40:11] WARNING: res_format_attr_siren7.c:52 siren7_parse_sdp_fmtp: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring. -- Executing [200@sets:2] Playback("PJSIP/0000f30B0B02-00000004", "hello-world") in new stack -- <PJSIP/0000f30B0B02-00000004> Playing 'hello-world.slin' (language 'en') > 0x7f6484017520 -- Strict RTP qualifying stream type: audio > 0x7f6484017520 -- Strict RTP switching source address to <EXTERNAL IP CLIENT>:4000 -- Executing [200@sets:3] Hangup("PJSIP/0000f30B0B02-00000004", "") in new stack == Spawn extension (sets, 200, 3) exited non-zero on 'PJSIP/0000f30B0B02-00000004'
I was wondering if you had any suggestions on which troubleshooting steps I should take or what logs with what log level I should be taking a look at to fix this issue. Thanks in advance for your answer!