[asterisk-users] Everyone is busy/congested: IP Trunk

[bilal ghayyad]

Dear Amit;

Special thanks for your greate help and support.

Sorry for delaying in reply, I was busy during this
week.

It worked with very poor and noise voice, and
disconnect after around 5 seconds, but it worked in
the direct mode (by using trustip=yes so Asterisk does
not register on the softswitch). But maybe this voice
problem was because of the network (I will explain my
network situation).

Before I explain my network situation, I would like to
know why in registering mode (by using register =>
directive and letting asterisk registering on the
softswitch), Asterisk was registering successfully but
call was not arrive for the softswitch (does not know
if Asterisk sent it or did not send it). The question:
is there a kind of packets negotiation during the SIP
registeration that determine the facility of call
exchaning? The softswitch ables to receive calls from
any SIP endpoint, why this does not do happen with
Asterisk if Asterisk registered? But it receive and
manipulate the calls if Asterisk work via trustip
(without registeration)?!! Actually, when Asterisk was
registering on the softswitch, I was see the
registeration on the softswitch, but I did not see
even the call attempt.

Regarding to my network status (that might be the
reason of having very poor and noise voice and
disconnecting the line after around 5 second),
actually the softswitch in public IP address and it is
located in Germany, while the Asterisk in Kuwait and
it is behind NAT (a private IP address), and the
softphone also have a private IP address (in the same
LAN with the Asterisk), so the softphone was
registering on the Asterisk, when the softphone send
the call for Asterisk then Asterisk was sending it for
the the softswitch in Germany via the SIP Trunk.

Do u think that because Asterisk Nated? In that case,
do u think the VPN will resolve the problem (VPN
between Asterisk network in Kuwait, and the Softswitch
network in Germany)? Or there is a settings should be
done?

Regards
Bilal


----------------------------
I have the same problem.

I trying with more 4 SIP providers, the account is
registering, receive
 
inboud calls, but can`t make outbound calls for
"congestion".

Can be the out call id the problem?

Thanks
Gabriel
----- Original Message ----- 
From: <joakimsen at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial
Discussion" 
<asterisk-users at lists.digium.com>
Sent: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is
busy/congested: IP Trunk


> No:
>
> register => abc:123 at xyz
>
> [peer]
> host=zzz
>
> Its possible to make mistakes and typos you know.
Maybe you can post
> your config file and we can help you.
>
> On 10/26/07, bilal ghayyad <bilmar_gh at yahoo.com>
wrote:
>> Hi Pablo;
>>
>> How the IP address will be wrong, and asterisk able
to
>> do registeration on the destination?
>>
>> If the IP address wrong, so I will not be able to
>> register on that IP address.
>>
>> Regards
>> Bilal
>>
>> > Hi List;
>>
>>
>> Ip address to destination?
>>
>> Unable to create channel of type SIP (cause 3 - No
>> route to destination)
>>
>> i think you have the wrong ip information
>>
>>
>>
>> >
>> > I established an SIP IP Trunk between Asterisk
and
>> > another softswitch (asterisk registered on the
>> > softswitch successfully) and I saw this on the
>> > softswitch.
>> >
>> > >From firefly softphone, I was need to do a call
to
>> be
>> > via this softswitch (ofcourse, the softphone will
>> send
>> > for asterisk and asterisk should route to the
>> > softswitch based on the extensions.conf
>> > configurations.
>> >
>> > But, always I receive this message (and the call
>> does
>> > not even reach to the softswitch, it is not
sended
>> > from Asterisk to the softswitch):
>> >
>> > Executing [9617565116 at EgyptInternationalVoIP:1]
>> > Dial("SIP/EgyptOeratorSIP-09f9bed0",
>> > "SIP/9617565116 at EgyptAlooNet") is new stack
>> >
>> > Unable to create channel of type SIP (cause 3 -
No
>> > route to destination)
>> >
>> > Everyone is busy/congested at this time (1:0/0/1)
>> >
>> > Anyone faced that?
>> >
>> > Is it related to a paramater that control number
of
>> > allowed channels per IP trunk? Maybe I have such
>> > parameters is 0 ? I do not know even if there is
>> such
>> > parameter.
>> >
>> > At the softswitch, I do not see even any attempt
>> > (nothing related to the dialed number), so why
>> > Asterisk does not send the called number to the
>> > softswitch and why asterisk assume there is not
>> > available channel?
>> >
>> > The softphone codec is g729a and the softswitch
>> > support such codec. Also, if it is a codec
matter,
>> > then call should be send to the softswitch, and
the
>> > softswitch will gives an error related to the
codec
>> > missmatch.
>> >
>> > Any help?
>> >
>> > Regards
>> > Bilal Ghayad


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