[HELP] 3 issues with PJSIP

Asterisk Asterisk Support
1

I have just started learning PJSIP, and at the moment I am running into to problems:

  1. I cannot get an endpoint that sits behind a NAT to open an RTP stream. I can register and initiate calls from the endpoint (a snom300), but when dialling out the endoint keeps giving a ringing tone, and never opens the audio stream. Below is my endpoint config:
[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============USER 101
[101]
type=endpoint
transport=simpletrans
context=internal
disallow=all
allow=g729
auth=auth101
aors = 101
direct_media=no
rtp_symmetric=yes
force_rport=yes

[101](aor-single-reg)

[auth101](auth-userpass)
password=snom300
username=101
  1. Sometimes I need to register an endpoint, but it’s previous registration is still active, which gives me the following message:
  1. How do i reload config changes to pjsip.conf without doing a full “core restart now”

Any help is appreciated.

2

I haven’t used pjsip, but:

  1. Doesn’t modules reload
    work?

  2. This sounds like the penalty for allowing multiple registrations. From the sample configuration, I would say:

; Modify the “max_contacts=” line to change how many unique registrations to allow.

  1. that suggest that Asterisk isn’t providing a valid contact address, or maybe it is only in ICE and the snom doesn’t understand that.