[VoIP] Sip, stupid sip, is biting me!
Dean Clark
dgclark at oldphoneco.com
Tue Feb 20 19:57:30 CST 2007
Here's a link explaining what I outlined:
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip
Dean Clark wrote:
> Sorry, I should have explained the entries a bit further. Correct on the
> externalrefresh=10 line, I believe that DNS is resolved every 10
> minutes. Also the localnet line is important as it defines the local
> subnet, of course this needs to be set to the actual local subnet not
> necessarily 192.168.1.0 as in my example. I'm using this on 1.2 and have
> also used it in a recent build of 1.4 but I have not had a chance to
> verify it.
>
> Dean
>
>
> Rusty Dekema wrote:
>
>> Originally, my question about this was going to be, "How would
>> Asterisk know when your IP address changes?"
>>
>> But, seeing your externrefresh=10 line, I am guessing this makes
>> Asterisk re-resolve your DNS name every 10 (seconds / minutes)?
>>
>> If so, that is very cool, and should also solve Jayson's problem. I
>> was not aware of this feature in Asterisk. Is it new as of 1.4, or
>> have you been using that in 1.2 as well?
>>
>> Rusty
>>
>>
>>
>> On 2/20/07, Dean Clark <dgclark at oldphoneco.com> wrote:
>>
>>
>>> 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
>>>>>>
>>>>>>
>>>>>> _______________________________________________
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>>>>>> Project Web Page: http://www.ckts.info/
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
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>>>>> Project Web Page: http://www.ckts.info/
>>>>>
>>>>>
>>>>>
>>>>>
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>>>>
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>
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