[OpenSIPS-Users] Questions about initial setup

Brett Nemeroff brett at nemeroff.com
Sat May 24 06:59:59 CEST 2014


Well the thing is.. if you are a registrar and users are registering to
your domain name like 1234 at domain.com.. Then your server needs to know it
is domain.com else when it hits this block (if you kept it from the
examples):

        if (!uri==myself) {
                append_hf("P-hint: outbound\r\n");
                route(relay);
        }

The call will attempt to be relayed.. and if the domain IS actually
pointing to yourself.. well SURPRISE, you have a loop again.

so "myself" means that the domain IS a valid alias..

(this is the short answer)

Remember, the LISTEN is what IP it'll get packets on.. regardless of domain
name.. used in the signaling.
-Brett



On Fri, May 23, 2014 at 8:07 PM, Kurtis Heimerl <kheimerl at endaga.com> wrote:

> Getting back on this horse, what do you mean "alias setup". There seem to
> be two different aliases: one for users (x at y -> 1000 at foo) and one for the
> service itself:
>
> Restarting opensips: opensipsListening on
>              udp: 192.168.0.0.1 [192.168.0.0.1]:5060
> Aliases:
>              udp: x.y.server.name:5060
>
> Which of these need to be configured for the server to work as a
> registrar? The former seems easy enough, but doesn't make a lot of sense
> architecturally. The second makes a lot more sense, but I can't find how to
> add another alias (that one is auto detected).
>
> Thanks again!
>
>
> On Thu, May 15, 2014 at 11:41 AM, Brett Nemeroff <brett at nemeroff.com>wrote:
>
>> So OpenSIPs doesn't do anything on it's own. If you just downloaded, and
>> compiled... I can say with like 99% certainty it's not going to do what you
>> want. It's just a toolkit.
>>
>> For example, if your config doesn't explicitly check the register to see
>> if it's valid, it's not doing it.
>>
>> The sample config i think has some of these bits littered around it..
>> Maybe you don't have your alias setup right in which case it would try to
>> relay it.
>>
>>
>> On Thu, May 15, 2014 at 12:46 PM, Kurtis Heimerl <kheimerl at endaga.com>wrote:
>>
>>> This makes *a lot* of sense. Thanks!I expected the SIP REGISTER to be
>>> terminated at the instance rather than forwarded as I had that user/pw
>>> combo in the subscribers table. I assume it doesn't do that by default I
>>> have to add a command to the routing logic to do that? Or, architecturally,
>>> should I just forward that request anyhow?
>>>
>>> Thanks!
>>>
>>>
>>> On Thu, May 15, 2014 at 2:11 AM, Brett Nemeroff <brett at nemeroff.com>wrote:
>>>
>>>> Normally this means that you haven't actually done anything with the
>>>> call, but you are t_relaying it out.
>>>>
>>>> In other words, the RURI was destined for your opensips box.. That's
>>>> how it got there.. it hit opensips, then you sent it back out.. But you
>>>> never adjusted the RURI to point to the next hop by either changing the
>>>> request user part or domain part. So it sends it back to itself (loopback)
>>>> until Max-Forwards expires and returns the 483.
>>>>
>>>> If you remove the 483 check, you get a massive loop on the loopback
>>>> interface until you start to kill server resources, which gave you the 408.
>>>>
>>>> Be sure you are actually doing some change to the message before you
>>>> send it back out. Try to xlog the r-uri before you t_relay() like this:
>>>>
>>>> xlog("L_INFO","Sending call out to $ru");
>>>>
>>>> If you watch sip traffic on your loopback interface, you'll see what
>>>> all the madness is about.. I don't really recommend it, since it very
>>>> quickly will consume your terminal, but you'll at least understand what
>>>> exactly is happening..
>>>>
>>>> Good luck!
>>>> -Brett
>>>>
>>>>
>>>>
>>>> On Wed, May 14, 2014 at 8:36 PM, Kurtis Heimerl <kheimerl at endaga.com>wrote:
>>>>
>>>>> Hey Users,
>>>>>
>>>>> I'm a developer with a lot of experience in FreeSWITCH trying out
>>>>> OpenSIPS for a larger, more central piece of infrastructure. I started
>>>>> investigating it a few days ago with the following use case in mind:
>>>>>
>>>>> FS instance with users attached.
>>>>> OpenSIPS in the cloud routing the FS communications to another cloud
>>>>> provider (e.g., Twilio).
>>>>>
>>>>> I've got FS set up with our OpenSIPS box as an external provider and a
>>>>> set username and password. That seems to be working. I set up OpenSIPS from
>>>>> the opensips APT repository (http://apt.opensips.org/) and version
>>>>> 11.1. It's running, configured with TEXTDB, and I inserted the username and
>>>>> password (after modifying the DB schema to allow null emails, which I guess
>>>>> is an ancient bug... http://sourceforge.net/p/openser/bugs/593/).
>>>>>
>>>>> The registration from FS is being rejected as a 483 "Too many hops"
>>>>> This is strange. I removed the 483 check in the opensips cfg and now it's a
>>>>> 408 Timeout. OpenSIPS itself it complaining in the following way. The
>>>>> actual log is also available, but too large for the mailing list.
>>>>>
>>>>> May 15 01:27:34 [26230] WARNING:core:fm_malloc: Not enough free
>>>>> memory, will atempt defragmentation
>>>>> May 15 01:27:34 [26230] ERROR:tm:relay_reply: no more share memory
>>>>> May 15 01:27:34 [26230] WARNING:core:fm_malloc: Not enough free
>>>>> memory, will atempt defragmentation
>>>>> May 15 01:27:34 [26230] ERROR:tm:_reply_light: failed to allocate
>>>>> shmem buffer
>>>>>
>>>>> Googling this error brings up a few hits on issues with a memory leak,
>>>>> but that can't be the case here as it is just started.
>>>>>
>>>>> Ideas? What am I doing wrong? Is this is right use case for OpenSIPS
>>>>> anyhow?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
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>>>>
>>>
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>>
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>
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