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Roger Johnsen

2005-Nov-28 15:30 UTC

[Asterisk-Users] Accepting Inbound SIP Connections

Hello,

We have an * server setup and are trying to take inbound SIP calls from
our provider.  According to the asterisk log, our box "sees" the call
come in, however, it never seems to route the call, rather gives a
"Congestion" message and the calling end is disconnected.  (I've
included the output from the log file below.)

One important note is that our SIP provider doesn't have the concept of
"registration" so we don't have a register line in our sip.conf
file.
I'm assuming this is the problem in that we need our Asterisk to accept
the incoming call and route it accordingly.  I'm assuming this is what
"allowguest" does in the sip.conf but according to the documentation,
it's turned on by default.

Can anyone give me some pointers?

Here is the output from the log file:

Connected to Asterisk 1.0.9 currently running on asterisk1 (pid = 1229)
Verbosity is at least 3
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, 8773036836, 2) exited non-zero
on 'SIP/5060-b5430440'
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-b5430440'
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, 8773036836, 2) exited non-zero
on 'SIP/5060-b5430440'
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-b5430440'


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Nir Simionovich - CTO

2005-Nov-28 15:49 UTC

head link

[Asterisk-Users] Accepting Inbound SIP Connections

Hi Roger,
 
  Can you please send a 'sip debug' output, so we can see the actual SIP
trace of the messages ?
 
Nir S

  _____  

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Roger Johnsen
Sent: Tuesday, November 29, 2005 12:30 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Accepting Inbound SIP Connections



Hello, 

We have an * server setup and are trying to take inbound SIP calls from our
provider.  According to the asterisk log, our box "sees" the call come
in,
however, it never seems to route the call, rather gives a "Congestion"
message and the calling end is disconnected.  (I've included the output from
the log file below.)

One important note is that our SIP provider doesn't have the concept of
"registration" so we don't have a register line in our sip.conf
file.  I'm
assuming this is the problem in that we need our Asterisk to accept the
incoming call and route it accordingly.  I'm assuming this is what
"allowguest" does in the sip.conf but according to the documentation,
it's
turned on by default.

Can anyone give me some pointers? 

Here is the output from the log file: 

Connected to Asterisk 1.0.9 currently running on asterisk1 (pid = 1229) 
Verbosity is at least 3 
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15 
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, 8773036836, 2) exited non-zero on
'SIP/5060-b5430440' 
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15 
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-b5430440' 
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15 
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, 8773036836, 2) exited non-zero on
'SIP/5060-b5430440' 
    -- Executing AbsoluteTimeout("SIP/5060-b5430440", "15")
in new stack
    -- Set Absolute Timeout to 15 
    -- Executing Congestion("SIP/5060-b5430440", "") in new
stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-b5430440' 


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Tony Davidson

2005-Nov-28 16:28 UTC

head link

[Asterisk-Users] Accepting Inbound SIP Connections

Suggest checking your extensions.conf file under [from-sip-external].


By default it often has:

;give external sip users congestion and hangup

exten => _.,1,AbsoluteTimeout(15)

exten => _.,2,Congestion

exten => _.,3,Hangup


The above lines should be commented out and the following added:

include => from-pstn


Hope that helps.

tony


Roger Johnsen wrote:
> Hello,
>
> We have an * server setup and are trying to take inbound SIP calls 
> from our provider.  According to the asterisk log, our box "sees"
the
> call come in, however, it never seems to route the call, rather gives 
> a "Congestion" message and the calling end is disconnected. 
(I've
> included the output from the log file below.)
>
> One important note is that our SIP provider doesn't have the concept 
> of "registration" so we don't have a register line in our
sip.conf
> file.  I'm assuming this is the problem in that we need our Asterisk 
> to accept the incoming call and route it accordingly.  I'm assuming 
> this is what "allowguest" does in the sip.conf but according to
the
> documentation, it's turned on by default.
>
> Can anyone give me some pointers?
>
> Here is the output from the log file:
>
> Connected to Asterisk 1.0.9 currently running on asterisk1 (pid = 1229)
> Verbosity is at least 3
>     -- Executing AbsoluteTimeout("SIP/5060-b5430440",
"15") in new stack
>     -- Set Absolute Timeout to 15
>     -- Executing Congestion("SIP/5060-b5430440", "") in
new stack
>   == Spawn extension (from-sip-external, 8773036836, 2) exited 
> non-zero on 'SIP/5060-b5430440'
>     -- Executing AbsoluteTimeout("SIP/5060-b5430440",
"15") in new stack
>     -- Set Absolute Timeout to 15
>     -- Executing Congestion("SIP/5060-b5430440", "") in
new stack
>   == Spawn extension (from-sip-external, h, 2) exited non-zero on 
> 'SIP/5060-b5430440'
>     -- Executing AbsoluteTimeout("SIP/5060-b5430440",
"15") in new stack
>     -- Set Absolute Timeout to 15
>     -- Executing Congestion("SIP/5060-b5430440", "") in
new stack
>   == Spawn extension (from-sip-external, 8773036836, 2) exited 
> non-zero on 'SIP/5060-b5430440'
>     -- Executing AbsoluteTimeout("SIP/5060-b5430440",
"15") in new stack
>     -- Set Absolute Timeout to 15
>     -- Executing Congestion("SIP/5060-b5430440", "") in
new stack
>   == Spawn extension (from-sip-external, h, 2) exited non-zero on 
> 'SIP/5060-b5430440'
>
>
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>
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asterisk users - Nov 2005 - Accepting Inbound SIP Connections

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