[VoIP] Sip, stupid sip, is biting me!

Dean Clark dgclark at oldphoneco.com
Tue Feb 20 19:18:45 CST 2007


Correct me if I'm wrong, but I have a dynamic host set up through Dyndns 
and have added the following to my sip.conf, it seems to have alleviated 
much of my sip-nat-firewall-no static IP headaches.

nat=yes
externhost=myhost.dyndns.com
localnet=192.168.1.0/255.255.255.0
externrefresh=10


Dean Clark





Rusty Dekema wrote:
> Ok, upon taking more than 10 seconds to think about it, I agree with
> Shane that the reinvite setting probably has nothing to do with your
> problem.
>
> If I were you, I would try this, even though it isn't a permanent
> solution. Find out what your dynamically allocated IP address is, and
> then enter that in the externip line in sip.conf under [general]. Then
> see if you get two-way audio.
>
> If this works, we will need to find a way to deal with your dynamic
> IP. First of all, what kind of Internet connection is this (cable,
> dsl, etc), and who is the provider? For example, the Charter cable
> Internet service at my father's house provides, in theory, a dynamic
> IP address, but it only changes once or twice a year. If yours is
> anything like this, you could hardcode the IP in sip.conf and change
> it whenever your IP changes.
>
> Rusty
>
>
>
>
> On 2/20/07, Jayson Smith <ratguy at bellsouth.net> wrote:
>   
>> Hi,
>> Rtpstart and rtpend are set to 10000 and 20000 respectively, and that's what
>> I've got forwarded. Putting "canreinvite=no" and "nat=yes" in the "general"
>> section of sip.conf didn't help. I'm assuming here that externip is for
>> specifying my external ip? If so, that may be needed, and this presents a
>> problem, since my IP is dynamic. I do have a domain name through No-ip which
>> gets updated whenever my IP changes, but I don't assume there's a way to
>> have Asterisk automatically determine what my external IP is based on that?
>> Thanks for all your help.
>> Jayson.
>>
>> ----- Original Message -----
>> From: "Shane Young" <voiptandem at shaneyoung.com>
>> To: <voip at ckts.info>
>> Sent: Tuesday, February 20, 2007 4:50 PM
>> Subject: Re: [VoIP] Sip, stupid sip, is biting me!
>>
>>
>>     
>>> The canreinvite really only applies to a call coming in SIP and going
>>> to something else SIP, such as a SIP phone.
>>>
>>> What I understand to be Jayson's problem is simply any call coming in
>>> to his system (and not going to another devices) is having one-way
>>> audio.
>>>
>>> Make sure the ports which are forwarded through the firewall match
>>> what is in the rtp.conf.
>>>
>>> For example:
>>> rtpstart=16384
>>> rtpend=32766
>>>
>>> You should be forwarding ports 16384 through 32766.  If not, the
>>> asterisk box and the far end might negotiate a port which is blocked
>>> by the firewall.
>>>
>>> You can also use ethereal or wireshark to see what's going on.
>>>
>>> Addtionally, check out the functionality for externip in your sip.conf
>>> and see if that is needed for your setup or not.
>>>
>>> --Shane
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Quoting Rusty Dekema <rdekema at gmail.com>:
>>>
>>>       
>>>> On 2/20/07, Jayson Smith <ratguy at bellsouth.net> wrote:
>>>>         
>>>>> Hello,
>>>>> Isn't that just for a peer definition in sip.conf though? The way I
>>>>>           
>> have it
>>     
>>>>> set up, Sipnumber has been told to call 622 at bluegrasspals.com to reach
>>>>>           
>> me.
>>     
>>>>> Currently I have that pointed at an extension that plays a rather long
>>>>> message, then runs the echo test to test audio. But anyone can call
>>>>> 622 at bluegrasspals.com. There's no way to say "canreinvite=no" or
>>>>>           
>> "nat=yes"
>>     
>>>>> for any and all random strangers that might call in, is there?
>>>>> Jayson.
>>>>>           
>>>> I am not positive this will work, but I would try putting
>>>> "canreinvite=no" under the [general] heading in sip.conf. I think the
>>>> setting will then take effect for anonymous callers.
>>>>
>>>> Rusty
>>>> _______________________________________________
>>>> VoIP mailing list
>>>> VoIP at ckts.info
>>>> http://lists.ckts.info/mailman/listinfo/voip
>>>> Project Web Page: http://www.ckts.info/
>>>>
>>>>         
>>> --Shane
>>> +1-821-7311 CNET
>>>
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>       
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>     
> _______________________________________________
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>
>
>   



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