[SR-Users] Kamailio realtime fails to forward register with Asterisk.

[imperium broadcast]

Hi Charles,  the sip trace was carried out on the asterisk server. I have
also disable the fw on both servers. I suspect  it might be network issue
but I have no idea what.
I have also set up another server running asterisk 11 same problem so I
know its not asterisk 1.8. I get no errors on either sever and logs don't
show any issues. Its just an odd issue thats got me baffled.

Regards
Mick
On 14 Sep 2013 09:08, "Charles Chance" <charles.chance at sipcentric.com>
wrote:

> Hi,
>
> Could be something network related, or firewall? I know you stated phones
> could register directly to Asterisk but if nothing is showing in Asterisk
> SIP log, then the requests are never getting to it.
>
> Was your trace performed on Kamailio server or Asterisk?
>
> Cheers,
>
> Charles
>
>
>
> On 13 September 2013 17:12, imperium broadcast <
> imperium.broadcast at gmail.com> wrote:
>
>> Hi all I have been trying to figure out where Im going wrong with
>> following this guide:
>>
>> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
>>
>> The only thing that is different is I am using Asterisk 1.8.22.0
>>
>> Kamailio works fine and I am able to register a phone to it but the
>> forward register fails to register with Asterisk.
>>
>>
>> Its like Asterisk doesn't see the request even running sip debug on the
>> Asterisk Console it doesn't show the register attempts. I have to use ngrep
>> to see any thing.
>>
>> Realtime works fine when I register the phone on Asterisk.
>>
>> Although I can get round the register using AGI I get the same issue when
>> passing a call to asterisk, Asterisk doesn't see the invite.
>>
>> Am I missing something so obvious :-/
>>
>>
>> This is what I have in sip.conf
>> "IP change to protect the innocent!"
>>
>> [kamailio-2]
>> type=peer
>> host=1.1.1.1
>> fromdomain=1.1.1.1
>> context=outgoing
>> insecure=invite,port
>> directmedia=nonat
>> qualify=yes
>> disallow=all
>> allow=all
>>
>> and this is the sip trace  for the register
>>
>>
>> U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080
>> REGISTER sip:172.16.0.110:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
>> To: sip:102 at 172.16.0.110.
>> From: sip:102 at 172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
>> CSeq: 10 REGISTER.
>> Call-ID: 4670378d1185f5b6-25000 at 127.0.0.1.
>> Content-Length: 0.
>> User-Agent: kamailio (3.3.1 (x86_64/linux)).
>> Contact: <sip:102 at 172.16.0.112:5060>.
>> Expires: 120.
>> .
>>
>>
>> U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080
>> REGISTER sip:172.16.0.110:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0.
>> To: sip:102 at 172.16.0.110.
>> From: sip:102 at 172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5.
>> CSeq: 10 REGISTER.
>> Call-ID: 4670378d1185f5b6-25000 at 127.0.0.1.
>> Content-Length: 0.
>> User-Agent: kamailio (3.3.1 (x86_64/linux)).
>> Contact: <sip:102 at 172.16.0.112:5060>.
>> Expires: 120.
>>
>> and I have this for my
>> Asterisk binds in kamamilio.cfg
>>
>> #!ifdef WITH_ASTERISK
>> asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address"
>> asterisk.bindport = "5080" desc "Asterisk Port"
>> kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address"
>> kamailio.bindport = "5060" desc "Kamailio Port"
>> #!endif
>>
>>
>> --
>> Regards
>> Mick
>>
>>
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>>
>
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