[Freeswitch-users] SIP Trunk with Private Static IP?
Rob Forman
rob4manhere at gmail.com
Fri Feb 26 05:10:49 PST 2010
Here's a simple sip ping script (sip_ping.pl) I found and like:
http://pastebin.freeswitch.org/12250
Good luck,
Rob
On Feb 26, 2010, at 6:52 AM, Steven Ayre wrote:
> A SIP 'ping' is not a ICMP ping...
>
> It works by sending a OPTIONS SIP request to the gateway, which then
> responds with 200 OK. It has the advantage of working even if ICMP is
> filtered by a firewall and testing whether the SIP server software is
> running, not just whether the server is online.
>
> Best Regards,
> -Steve
>
>
> On 25 February 2010 15:02, Roly Maz <rm at callrica.co.za> wrote:
>> Many thanks for your prompt reply and the help
>>
>> I removed the LAN GW and kept the WAN GW.
>>
>> I have modified the standard internal and external sip profiles
>> accordingly
>>
>> What is odd is that if i run a ping from the windows command line,
>> I get a
>> reply from the SIP Server. However, if I setup a ping within FS, it
>> fails.
>>
>> I am investigating...
>>
>>
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Steven
>> Ayre
>> Sent: 25 February 2010 10:59 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] SIP Trunk with Private Static IP?
>>
>> Gateways do not require usernames and passwords. You are required to
>> set the parameter, but if no authentication is needed they are
>> ignored
>> so you can put anything in the field, so that is not a reason to
>> avoid
>> them.
>>
>> -Steve
>>
>>
>> On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
>>> Create two SIP profiles, each bound to one of your local IPs.
>>>
>>> You may create a gateway on the profile for the SIP trunk IP for the
>>> 10.42.0.1 server, but this is optional.
>>>
>>> You can then bridge calls via the SIP server using one of:
>>> <action application="bridge" data="sofia/gateway/gatewayname"/>
>>> <action application="bridge" data="sofia/profilename/number at 10.42.0.1
>>> "/>
>>>
>>> The advantages of using a gateway are:
>>> - supports authentication
>>> - will monitor the gateway to detect if it goes down (so calls fail
>>> instantly rather than after a timeout)
>>>
>>> As for the default gateway, it is the IP you send via to reach IPs
>>> that are not on a network you are connected directly to - you should
>>> probably only have one set, and it should be the one you go via to
>>> reach the Internet.
>>>
>>> -Steve
>>>
>>>
>>> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>>>> Hi Community
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> My Provider provides the following info when they supply a SIP
>>>> trunk:
>>>>
>>>>
>>>>
>>>> · A direct connection into their network. i.e. they provide
>> private
>>>> IPs:
>>>>
>>>> · An IP address I must use for my FS box e.g. IP: 10.
>>>> 42.0.66
>> MASK:
>>>> 255.255.255.248 GW: 10. 42.0.68
>>>>
>>>> · An IP address for their SIP server 10.42.0.1
>>>>
>>>>
>>>>
>>>> I have setup a dual homed FS box (Windows Server 2008, latest FS
>>>> version)
>>>>
>>>>
>>>>
>>>> NIC 1 – Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW:
>>>> 10.0.2.253
>>>>
>>>> NIC 2 – SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248
>>>> GW: 10.
>>>> 42.0.68
>>>>
>>>>
>>>>
>>>> Windows complains about multiple gateways – which I ignore? I can
>>>> ping
>>>> internal addresses and the SIP Server
>>>>
>>>>
>>>>
>>>> When I fire up FS, I can register Xlite phones on my LAN. I can
>>>> dial and
>>>> hear the test IVR (5000)
>>>>
>>>>
>>>>
>>>> This means my Internal SIP Profile is ok.
>>>>
>>>>
>>>>
>>>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>>>
>>>>
>>>>
>>>> Creating a gateway doesn’t make sense, because I am not supplied a
>>>> username/password?
>>>>
>>>>
>>>>
>>>> Any pointers would be most appreciated, I am sure I am missing
>>>> something
>>>> really simple.
>>>>
>>>>
>>>>
>>>> Roland
>>>>
>>>>
>>>>
>>>>
>>>>
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>>>>
>>>>
>>>
>>
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