[solved] Asterisk 13 & PJSIP - no sound and doesn't hang up

Asterisk Asterisk Support
1

Finally got Asterisk 13.6 and pjsip built and running together nicely without any apparent errors.

Except, no sound and the clients don’t hangup at the end.

It doesn’t matter how I try and make a sound, I just get silence. No NAT or firewall that I know of.

pjsip.conf

;===============TRANSPORT

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============EXTENSION 6001

[6001]
type=endpoint
context=internal
disallow=all
allow=alaw
transport=simpletrans
auth=auth6001
aors=6001

[auth6001]
type=auth
auth_type=userpass
password=xxxxxxx
username=6001

[6001]
type=aor
max_contacts=3

extensions.conf

[internal] exten => 100,1,Answer() same => n,Wait(5) ;same => n,Playback(en_GB/weasels-eaten-phonesys) same => n,SayDigits(123) same => n,Wait(5) same => n,Hangup()

Console output

Connected to Asterisk 13.6.0 currently running on Asterisk2015 (pid = 682) -- Executing [100@internal:1] Answer("PJSIP/6001-00000002", "") in new stack > 0x32db210 -- Probation passed - setting RTP source address to x.x.x.x:4002 -- Executing [100@internal:2] Wait("PJSIP/6001-00000002", "5") in new stack -- Executing [100@internal:3] SayDigits("PJSIP/6001-00000002", "123") in new stack -- Executing [100@internal:4] Wait("PJSIP/6001-00000002", "5") in new stack -- Executing [100@internal:5] Hangup("PJSIP/6001-00000002", "") in new stack == Spawn extension (internal, 100, 5) exited non-zero on 'PJSIP/6001-00000002'

2

Did you solve this?

3

My apologies! Yes, I did, and I meant to come back here to answer it.

Basically, it looks like some kind of ISP-related firewall thing that I cannot change in the router they provide. After much scratching of head, the answer is to use “Allow IP rewrite” with MicroSIP.
I never managed to find the equivalent setting with Zoiper, though.

But MicroSIP is working for me, and that’ll do!