[Asterisk-Users] Test settings

[Ronald Wiplinger]

I should connect to a gateway and got following info:

Username = Password = NONE    (not very secure!!!)
SIP
port 5060
IP address
For a trunk line dial 1234 and continue the number you want to reach at 
PSTN.
codex g723   (I guess it should be g723.1)


vpbx*CLI>
    -- Executing NoOp("SIP/615-127a", "SIP/12340939775516 at sip-xxxx") in 
new stack
Apr  8 23:06:45 NOTICE[12235]: rtp.c:451 ast_rtp_read: RTP: Received 
packet with bad UDP checksum
    -- Timeout on SIP/615-127a
  == CDR updated on SIP/615-127a
    -- Executing Goto("SIP/615-127a", "#|1") in new stack
    -- Goto (default,#,1)
    -- Executing Playback("SIP/615-127a", "demo-thanks") in new stack
    -- Playing 'demo-thanks' (language 'en')
    -- Executing Hangup("SIP/615-127a", "") in new stack
  == Spawn extension (default, #, 2) exited non-zero on 'SIP/615-127a'
    -- Executing Hangup("SIP/615-127a", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/615-127a'



extensions.conf

exten => _1234.,1,NoOP(SIP/${EXTEN}@sip-xxxx);
exten => _1234.,1,Dial(SIP/${EXTEN}@sip-xxxx);


[sip-xxxx]                ; test gw
type=peer               
host=22.22.11.42
context=inhouse
nat=yes
canreinvite=no
insecure=very
dtmfmode=inband
disallow=all
allow=g723.1
qualify=yes



Q:
1. What triggers:    RTP: Received packet with bad UDP checksum
2. How can I solve that?


bye

Ronald