When dialing local extension phone rings, but when picked up no sound


(Ojwhaaa) #1

When I call for example from extension 204 to 207, the phone rings but when I pick it up no sound.
When I hang up the other phone doesn’t seem to know this either, what’s going on??
And yes, I configured local network on Asterisk sip settings

I’m using FreePBX 15.0.16.42.

The same thing for the sip trunk, when I call from the outside the phone also rings but no sound at all.

15845 [2020-12-12 19:29:18] VERBOSE[21934][C-00000024] app_while.c: Jumping to priority 13
15846 [2020-12-12 19:29:18] VERBOSE[21934][C-00000024] pbx.c: Executing [s@func-apply-sipheaders:14] Return(“PJSIP/207-00000039”, “”) in new stack
15847 [2020-12-12 19:29:18] VERBOSE[21934][C-00000024] app_stack.c: Spawn extension (from-internal, 207, 1) exited non-zero on ‘PJSIP/207-00000039’
15848 [2020-12-12 19:29:18] VERBOSE[21934][C-00000024] app_stack.c: PJSIP/207-00000039 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
15849 [2020-12-12 19:29:18] VERBOSE[21934][C-00000024] app_dial.c: Called PJSIP/207/sip:207@10.10.10.141:5060
15850 [2020-12-12 19:29:18] VERBOSE[22065] netsock2.c: Using SIP RTP Audio TOS bits 184
15851 [2020-12-12 19:29:18] VERBOSE[22065] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
15852 [2020-12-12 19:29:18] VERBOSE[22065] netsock2.c: Using SIP RTP Audio CoS mark 5
15853 [2020-12-12 19:29:19] VERBOSE[21934][C-00000024] app_dial.c: PJSIP/207-00000039 is ringing
15854 [2020-12-12 19:29:19] VERBOSE[21934][C-00000024] app_dial.c: PJSIP/207-00000039 is ringing
15855 [2020-12-12 19:29:25] VERBOSE[21934][C-00000024] app_dial.c: PJSIP/207-00000039 answered PJSIP/204-00000038
15856 [2020-12-12 19:29:25] VERBOSE[21951][C-00000024] bridge_channel.c: Channel PJSIP/207-00000039 joined ‘simple_bridge’ basic-bridge <5637ffec-3e14-46ba-9b4f-1860c065e4a6>
15857 [2020-12-12 19:29:25] VERBOSE[21934][C-00000024] bridge_channel.c: Channel PJSIP/204-00000038 joined ‘simple_bridge’ basic-bridge <5637ffec-3e14-46ba-9b4f-1860c065e4a6>
15858 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] bridge_channel.c: Channel PJSIP/204-00000038 left ‘simple_bridge’ basic-bridge <5637ffec-3e14-46ba-9b4f-1860c065e4a6>
15859 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] app_macro.c: Spawn extension (macro-dial-one, s, 56) exited non-zero on ‘PJSIP/204-00000038’ in macro ‘dial-one’
15860 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/204-00000038’ in macro ‘exten-vm’
15861 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Spawn extension (ext-local, 207, 2) exited non-zero on ‘PJSIP/204-00000038’
15862 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [h@ext-local:1] Macro(“PJSIP/204-00000038”, “hangupcall,”) in new stack
15863 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/204-00000038”, “1?theend”) in new stack
15864 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx_builtins.c: Goto (macro-hangupcall,s,3)
15865 [2020-12-12 19:29:57] VERBOSE[21951][C-00000024] bridge_channel.c: Channel PJSIP/207-00000039 left ‘simple_bridge’ basic-bridge <5637ffec-3e14-46ba-9b4f-1860c065e4a6>
15866 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/204-00000038”, “0?Set(CDR(recordingfile)=)”) in new stack
15867 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/204-00000038”, "PJSIP/207-00000039 montior file= ") in new stack
15868 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/204-00000038”, “1?skipagi”) in new stack
15869 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx_builtins.c: Goto (macro-hangupcall,s,7)
15870 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/204-00000038”, “”) in new stack
15871 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/204-00000038’ in macro ‘hangupcall’
15872 [2020-12-12 19:29:57] VERBOSE[21934][C-00000024] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/204-00000038’


#2

Go to Sip Setting>Local Networks change this setting to “192.168.0.0/16”. match this to your networks. Using a “nn.nn.0.0/16”. I Answer to this to a similar problem in
https://community.freepbx.org/t/no-audio-after-answering/63315


(Ojwhaaa) #3

Thanks for the aswer, but unfortunately it doesn’ t work, I installed also the softphone wave lite on two iphones, exactly the same problem. And yes, nat=yes and my local network I set to 10.10.0.0/16.

The phones are registerring and are ringing when I call the extension, but no sound at all.
What am I doing wrong here?


#4

That’s a chan_sip setting, but your extensions are pjsip?

Confirm that you have restarted (not just reloaded) Asterisk after changing this setting.

If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
(or sip set debug on if the extensions really are chan_sip)
make a failing call, paste the Asterisk log (which will now include a SIP trace) at pastebin.freepbx.org and post the link here.