[SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1

[Barry Flanagan]

On 30 May 2013 21:09, Michael Leuker <michael at leuker.me> wrote:

> Thank you so much for pointing me in the right direction! It was the
> missing alias. Now the extensions are working, but there's one more
> problem: When I call the PBX (echo-test) or from one extension to another,
> I get a hangup because Asterisk doesn't seem to receive an ACK from the
> client:
>
> "Retransmission timeout reached on transmission
> OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA for seqno 2 (Critical Response)"
>
>
Solution is going to lie in the kamailio logs I think:

"Retransmitting #5 (no NAT) to 198.23.139.21:5060: "


...for whatever reason, Kamailio is not replying to these.

-Barry Flanagan


I have attached the log for two local calls that hangup after about 30s and
> the sequence for an incoming call from one of the trunks that works without
> problems. So you can map all the IPs:
>
> --------
> 198.23.139.21 (5060): Kamailio
> 198.23.139.21 (5080): Asterisk
> --------
> 192.168.178.33: Home network local IP
> 188.105.112.187: Home network external IP
> --------
> 94.75.247.45: Trunk
> --------
>
> NAT was set to "No - RFC3581" for these captures, but I've tried all other
> possibilities (including nathelper / rtpproxy) without success. Do you (or
> anybody else) have any idea where to look in order to solve this problem?
>
>
>
> On Thu, May 30, 2013 at 10:36 AM, Barry Flanagan <barry at flanagan.ie>wrote:
>
>> On 29 May 2013 19:23, Michael Leuker <michael at leuker.me> wrote:
>>
>>> Sure, here's the sequence for an inbound call via the "LPhone" trunk
>>> that was supposed to go through to extension 1001. The extension was set to
>>> "NAT" in the FreePBX settings. Just ask if you need more background.
>>>
>>>
>> The Asterisk part looks fine. It is sending the call to
>> 1001 at 198.23.139.21:5060 which I presume is your Kamailio instance.
>>
>> for some reason Kamailio is not recognising the user. Could it be that
>> you do not have the ip 198.23.139.21 set up on Kamailio as an alias, or
>> that the user 1001 is registering to a different domain?
>>
>> Kamailio would be looking in the location table for "username='1001' AND
>> domain='198.23.139.21'"
>>
>> You should check the Kamailio logs for what is happening when Asterisk
>> sends it the INVITE for 1001 at 198.23.139.21
>>
>> Hope this helps.
>>
>> -Barry
>>
>>
>>> On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan <barry at flanagan.ie>wrote:
>>>
>>>> On 29 May 2013 10:25, Michael Leuker <michael at leuker.me> wrote:
>>>>
>>>>> Thank you very much for sharing your insights, Barry! I am facing the
>>>>> same problem that Trevor described:
>>>>>
>>>>> Things are working just fine on their own, but as soon as FreePBX
>>>>> comes into play, calling extensions becomes impossible because of the
>>>>> different tables used. Removing the password from FreePBX (and setting the
>>>>> Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but
>>>>> even though the extension shows as registered in FreePBX, it always shows
>>>>> as busy:
>>>>>
>>>>> chan_sip.c:23237 handle_response_invite: Failed to authenticate on
>>>>> INVITE to '"xxxxxxxx" <sip:xxxxxxxx at 198.23.139.21>;tag=as72a4117a'
>>>>>     -- SIP/1001-00000006 is circuit-busy
>>>>>
>>>>>
>>>> Can you do "sip set debug on" on Asterisk and make a call and  post the
>>>> output?
>>>>
>>>> -Barry
>>>>
>>>>
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>>>
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>>
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>
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