[OpenSIPS-Users] Questions about initial setup

Fri May 30 06:20:54 CEST 2014

Just wanted say that the "alias=" line worked perfectly. Thanks so much for
your help brett.

On Fri, May 23, 2014 at 11:58 PM, Brett Nemeroff <brett at nemeroff.com> wrote:

> Try just:
>
> alias=domain.com
>
> http://www.opensips.org/Documentation/Script-CoreParameters-1-9#toc26
>
> Note that there are some tricks you normally have to do to make EC2 work.
> You are not learning the easy way :)
>
> Check out this thread for some pointers:
> http://lists.opensips.org/pipermail/users/2010-April/012367.html
>
>
>
> On Sat, May 24, 2014 at 12:45 AM, Kurtis Heimerl <kheimerl at endaga.com>
> wrote:
>
>> This totally makes sense! Thanks so much. Due to the intricacies of our
>> design (ec2 in action) the automatic alias detection isn't working (it gets
>> the name from the local DNS which isn't routable externally). How do I get
>> it to know that "myself" is a certain domain name? I set SIP_DOMAIN in the
>> configuration, that didn't seem to change anything.
>>
>> Thanks again!
>>
>>
>> On Fri, May 23, 2014 at 9:59 PM, Brett Nemeroff <brett at nemeroff.com>
>> wrote:
>>
>>> 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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>>>>
>>>
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>>
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>
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