[asterisk-users] NAT issue (i think?)

Mon Oct 4 03:06:44 CDT 2010

sorry for the late reply. got tied with something else. please see 
attached ngrep result.

my setup is simple, i add the user on the realtime database. i configure 
a phone to set my domain (i have DNS SRV enabled on my domain)

my clients are usually using ADSL links and have those SOHO ADSL router 
(usually linksys). IP Phones are on DHCP, i set NAT Keep-Alive to Yes 
and NAT Mapping to Yes, i also set the ffg, to Yes, Handle VIA received, 
Handle VIA rport,Insert VIA received and Insert VIA rport.

also if there are more than 1 phone behind the same NAT, i set different 
SIP port and each phone, i'm just wondering why it only happens on 
linksys phones, using yealink and grandstream it's ok.

Thanks again.

Regards
Ron

On 9/29/10 7:39 AM, Danny Dias wrote:
> Hello Ron..
>
> The answer that i see here is only a trying to a Register...means the
> REGISTRATION procedures are taking a significant amount of time.
>
> You should get a 200 OK
>
> Can you lease make a simple draw of your architecture? seems to be a NAT
> problem, that's for sure
>
> REgards!
>
> 2010/9/28 Ron<nhadie at gmail.com>
>
>> Hi Danny
>>
>> On the pap2 by default it is set to 3600 and i have not change that.
>> by the way, is the NAT keep-alive same with the NOTIFY message? coz i am
>> seeing my asterisk respond to those as bad event could that be causing
>> it to loose the registration?
>>
>> here's the registration from ngrep:
>>
>> U 78.65.34.12:5094 ->  12.34.56.78:5060
>> REGISTER sip:sip.mydomain.com SIP/2.0.
>> Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport.
>> From: Kristine<sip:456789 at sip.mydomain.com<sip%3A456789 at sip.mydomain.com>
>>> ;tag=68fc368d164925e0o0.
>> To: Kristine<sip:456789 at sip.mydomain.com<sip%3A456789 at sip.mydomain.com>
>>> .
>> Call-ID: c9bd8b57-f7bdc794 at 192.168.1.52.
>> CSeq: 116228 REGISTER.
>> Max-Forwards: 70.
>> Contact: Kristine<sip:456789 at 78.65.34.12:5094>;expires=3600.
>> User-Agent: Linksys/PAP2T-3.1.15(LS).
>> Content-Length: 0.
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
>> Supported: x-sipura.
>> .
>>
>>
>> U 12.34.56.78:5060 ->  78.65.34.12:5094
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP
>> 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094.
>> From: Kristine<sip:456789 at sip.mydomain.com<sip%3A456789 at sip.mydomain.com>
>>> ;tag=68fc368d164925e0o0.
>> To: Kristine<sip:456789 at sip.mydomain.com<sip%3A456789 at sip.mydomain.com>
>>> .
>> Call-ID: c9bd8b57-f7bdc794 at 192.168.1.52.
>> CSeq: 116228 REGISTER.
>> User-Agent: Asterisk PBX.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
>> Supported: replaces.
>> Content-Length: 0.
>>
>>
>> On 9/28/10 7:24 PM, Danny Dias wrote:
>>> You have to increase the time of expiration for the Register...on linksys
>>> devices is located on Proxy and Registration section under the EXTN:
>> (Where
>>> N is the extension number)
>>>
>>> Try putting this to: 3600
>>>
>>> To check wheter or not is loosing Register, try with ngrep-sip and check
>> it:
>>>
>>> ngrep -p -q -W byline port 5060>register.pkt
>>>
>>> Then post here the content of register.pkt; but please, after issuing the
>>> change explained above!
>>>
>>> Regards!
>>>
>>> 2010/9/28 Ron<nhadie at gmail.com>
>>>
>>>> Hi All.
>>>>
>>>> got this problem that IP phones could not re-register to my server. even
>>>> if device is power cycled it still would not register. the solution i
>>>> found was to change the SIP port settings on the phone and it will
>>>> register. but after registration expires and its time to re-register the
>>>> same thing will happen, so i have to update the port settings again just
>>>> to make it work which is troublesome.
>>>>
>>>> i'm using Asterisk 1.4.31 with the following realtime config:
>>>>
>>>> rtcachefriends=yes
>>>> rtsavesysname=yes
>>>> rtupdate=yes
>>>> rtautoclear=no
>>>>
>>>> one thing i noticed is that it only seems to happen on linksys devices
>>>> e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no
>>>> client has complain about it.
>>>>
>>>> hope anyone can help. thank you.
>>>>
>>>> regards
>>>> Ron
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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