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: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk
> No:
>
> register => abc:[EMAIL PROTECTED]
>
> [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 <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED]:1]
>> > Dial("SIP/EgyptOeratorSIP-09f9bed0",
>> > "SIP/[EMAIL PROTECTED]") 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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