[Asterisk-Users] Re: I never get to hear more than 5s of the demo channels

[Jean-Yves Avenard]

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Dear all.

I'm new to this so please forgive my ignorance if I missed something 
obvious.

I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not 
linux but that's all we have available at that stage).
After some struggle to understand how everything works, I set up some 
SIP accounts for test purposes.


I can log in, make calls to some of the demo system (1234, 1000 etc...) 
but the playback will always stop after 5s. I mean: I *hear* something 
(a lady) and after 5 s it stops, and X-lite displays: hung-up

On Asterisk console I get the following messages:

*CLI> Jun 28 08:41:42 NOTICE[135336960]: chan_sip.c:4933 
handle_response: Peer 'avenardj' is now REACHABLE!
     -- Executing Goto("SIP/avenardj-acfc", "default|s|1") in new stack
     -- Goto (default,s,1)
     -- Executing Wait("SIP/avenardj-acfc", "1") in new stack
     -- Executing Answer("SIP/avenardj-acfc", "") in new stack
     -- Executing DigitTimeout("SIP/avenardj-acfc", "5") in new stack
     -- Set Digit Timeout to 5
     -- Executing ResponseTimeout("SIP/avenardj-acfc", "10") in new stack
     -- Set Response Timeout to 10
     -- Executing BackGround("SIP/avenardj-acfc", "demo-congrats") in 
new stack
     -- Playing 'demo-congrats' (language 'en')
Jun 28 08:41:53 NOTICE[135433216]: sched.c:218 sched_settime: Request 
to schedule in the past?!?!
Jun 28 08:41:56 WARNING[135336960]: chan_sip.c:498 retrans_pkt: Maximum 
retries exceeded on call 
2AAAB3C2-C88B-11D8-9F79-000D93AD5C52 at 192.168.2.3 for seqno 25040 
(Response)
   == Spawn extension (default, s, 5) exited non-zero on 
'SIP/avenardj-acfc'


I'm trying to connect to the SIP gateway over NAT from my home account.
Even without NAT when connecting over internet it will not exceed this 
5s time limit.

It works fine on the local network. I've looked for previous solution 
and it seems that each time somebody complained about such issue it was 
related to BSD system.
so is asterisk fully working on BSD? If you had this issue in the past 
; how did you resolve it?

Here is a sample of the sip.conf file for my username:

[user1]
type=friend
nat=yes                         ; phone may be behing nat
host=dynamic
reinvite=no
canreinvite=no
qualify=1000                    ; send udp every now and then to keep 
nat open
mailbox=101                     ; mailbox number
username=user1               ; username used for identification
secret=xxxxx                   ; password for registration
dtmfmode=info                   ; DTMF mode
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=sip


Also, as a side note. Some people mentioned that they didn't have such 
issue when the used SER as the SIP proxy ; is it possible to run SER 
and Asterisk on the same machine?

Any ideas? Help please !!!

Regards
Jean-Yves

- ---
Jean-Yves Avenard
Hydrix Pty Ltd - Embedding the net
www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724
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