[Asterisk-Users] getting asterisk to reliably answer a voip line

[jnuoiqweahf kajhdsff]

--- picciuX wrote:
> maybe you can try to issue a "sip show registry" on
> the console on a regular
> basis and watch if your * loose registration.

Ok:

asterisk1*CLI> sip show registry
Host                            Username       Refresh
State
proxy01.sipphone.com:5060       17476510045        105
Registered

Also, at my.sipphone.com, when I log in and view
advanced features, in "SIP Registrations" the status
is always "on line" and "Public IP address" shows the
IP address of the NAT device which my asterisk machine
is behind, followed by e.g. "(expires in 1020
seconds)".

According to both asterisk and sipphone, I'm never
losing registration.

> You can also turn on sip debug on the console, to
> see if the "unanswered
> calls" effectively reach asterisk or not.
I did "sip debug" on the console and got "SIP
Debugging enabled". Now, every 20 seconds or so, I
get:

<-- SIP read from 192.168.3.22:5060:

--- (0 headers 0 lines) Nat keepalive ---

Trying to call after enabling debugging, some calls
succeed and some fail (as usual), and I get no
indication of the call attempts on the console when
the call fails.
Every minute or so, I get long spiels on the console
(unrelated to the timing of my call attempts) starting
with:

REGISTER 13 headers, 0 lines
 Reliably Transmitting (no NAT) to
198.65.166.131:5060:
REGISTER sip:proxy01.sipphone.com SIP/2.0

which sometimes contain strange things like:

Destroying call
'8437bf4312a31d5432fb1bd6a3b53e1ac at proxy01.sipphone.com'
asterisk1*CLI>
<-- SIP read from 192.168.3.22:5060:

--- (0 headers 0 lines) Nat keepalive ---
asterisk1*CLI>
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.23:5060;branch=z5lR3bK653e4bb;rport=1937;received
=72.171.58.49
From:
<sip:17476510045 at proxy01.sipphone.com>;tag=as741dda96
To:
<sip:17476510045 at proxy01.sipphone.com>;tag=21a68532c2cd5d9b34affe6bba40a2e.
1bb5
Call-ID:
6418bf2376d9872efb1bd6a4b32f1ac at proxy01.sipphone.com
CSeq: 166 REGISTER
P-Behind-NAT: Yes
Contact: <sip:s at 72.171.58.49:1936>;q=0.00;expires=26
Contact: <sip:s at 72.171.58.49:1937>;q=0.00;expires=120
Content-Length: 0

--- (10 headers 0 lines)---
 Scheduling destruction of call
'723987f98ea31e2342fb1bd6a4b32e1bd at proxy01.sippho
ne.com' in 32000 ms

Which is strange because I had no incoming or outgoing
calls or call attempts at the time I got those
messages on the console, yet asterisk is talking about
destroying calls.

> In the
> latter, is sipphone that
> loose your registration,
Yes, this appears to be the case.

> so you maybe can lower the
> time before registration
> renewals.
But during the time I was doing tests and recording
the above information to put in this message, I had
several call attempts succeed and several fail, and
several minutes later, the SIP registration I
mentioned at my.sipphone.com was down from 1020
seconds to 681 seconds, and then later I checked again
and it was down to 412 seconds, etc. So all the while
when I was having some calls succeed and some fail, my
sipphone registration had not yet been renewed
(according to sipphone). So I don't know what all the
registration stuff is that asterisk is dumping to the
console in debug mode, but it's apparently not
reregistration with sipphone, since sipphone's timer
doesn't get reset by it, and it doesn't seem to have
any relationship to whether my incoming call attempts
succeed or fail.

> And turn on "qualify=yes" for your peer to
> keep fresh nat mappings
> on the router.
I tried that yesterday, and it seemed to have no
effect.

Based on this information, can you give any clue as to
what the problem might be?


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam
protection around 
http://mail.yahoo.com 

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com