[OpenSIPS-Users] OpenSIPS and Asterisk integration
sjs205
sjs205.linux at gmail.com
Sun Apr 14 16:20:04 CEST 2013
Hello Olle,
I had previously tried this, and it seemed to help somewhat, although I
then have to create the following fields too otherwise Asterisk
complains about not being able to update them:
alter table subscriber add column `fullcontact` int(35) DEFAULT NULL;
alter table subscriber add column `lastms` int(11) DEFAULT NULL;
alter table subscriber add column `useragent` int(11) DEFAULT NULL;
alter table subscriber add column `host` varchar(40) DEFAULT NULL,
alter table subscriber add column `port` varchar(5) DEFAULT `5060`;
alter table subscriber add column `regserver` varchar(40) DEFAULT NULL;
alter table subscriber add column `ipaddr` varchar(40) DEFAULT NULL;
alter table subscriber add column `defaultip` varchar(40) DEFAULT NULL;
alter table subscriber add column `regseconds` int(11) DEFAULT NULL;
But this doesn't help either since now when Asterisk updates the tables
I end up getting the following error when I try to register the user:
[Apr 14 13:37:33] WARNING[11455]: chan_sip.c:13880
__set_address_from_contact: Invalid contact uri 0 (missing sip: or
sips:), attempting to use anyway
[Apr 14 13:37:33] WARNING[11444]: chan_sip.c:13880
__set_address_from_contact: Invalid contact uri 0 (missing sip: or
sips:), attempting to use anyway
All in all, I'm not having much luck with that tutorial at all. I am
guess that the problem could be down to not having exact version numbers
for Asterisk and openSIPS, although the tutorial states "This tutorial
is made for OpenSIPS 1.8.x and Asterisk 1.8.x"
Another point that I've just realised from the tutorial - And this is
something I hadn't realised previously - is that openSIPS should be
dealing with the registration, and not passing this onto Asterisk. From
the routing section of opensips.cfg:
# requests for my domain
if (is_method("REGISTER")) {
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
}
if (!check_to()) {
send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
So in this instance, asterisk should only be dealing with the endpoints,
such as voicemail and stuff and sipusers is only shared with asterisk
because it needs to know which users should have voicemail, in which
case, it really shouldn't matter if the users do not have "host=dynamic"
set. Is that correct? If that is so, it would lead me to believe that
there is some issue with the configuration of openSIPS rather than
asterisk. Can anyone advise how I could debug this?
Kind regards,
Steven
On 04/14/2013 08:59 AM, Olle E. Johansson wrote:
> 13 apr 2013 kl. 22:08 skrev sjs205 <sjs205.linux at gmail.com>:
>
>> Hello N,
>>
>> Thanks for getting back to me on this. This is one of the issues with this tutorial, one can not set the asterisk sip_peers to dynamic since the tutorial creates a view from the 'subscriber' table and uses the 'domain' field as the 'host', 'default', 'ipaddr', 'fromdomain' and 'regserver' fileds in the asterisk table. And hence, if one changes the one of the user's 'host' field to dynamic in the database, that then changes all of these fields.
>>
>> So changing each of the sip_peers (or sipusers in this case) in the table to is not possible. Based on this, I would expect that this is an error with the tutorial itself, although I'm just not 100% sure.
> If you add a column called host to the subscriber table I am pretty sure, but not 100%, that OpenSIPS won't mind. Asterisk will be much more happy.
>
> /O
>> Steven
>>
>>
>> On 04/13/2013 08:43 PM, Nick Khamis wrote:
>>> Make sure that you have host=dynamic on both the general level (i.e.,
>>> sip.conf) and at the
>>> peer level (i.e., extensions, sip_peers in the database etc...)
>>>
>>>
>>> N.
>>>
>>> On 4/13/13, sjs205 <sjs205.linux at gmail.com> wrote:
>>>> Hello all,
>>>>
>>>> I'm going round and round in circles trying to integrate openSIPS and
>>>> asterisk using the tutorial found at:
>>>>
>>>> http://www.opensips.org/Resources/DocsTutAsterisk18
>>>>
>>>> I have managed to get openSIPS installed and starting without errors
>>>> using the configuration scripts included in the above tutorial. I have
>>>> also installed Asterisk 1.8 from the ubuntu repo and have managed to get
>>>> this working to.
>>>>
>>>> The problem comes when I follow the final section of the tutorial - 1.9
>>>> How to use it.
>>>>
>>>> I can do everything up to the "Test Voicemail" section where I am trying
>>>> to register my user using twinkle on a separate host. - I've added the
>>>> following to my /etc/hosts file on the twinkle client machine:
>>>>
>>>> 192.168.5.20 test.com test.com
>>>>
>>>> 192...20 is the openSIPS server, with "test.com" being the domain
>>>> assigned to the new sip user, alice, created. I know that this resolves
>>>> perfectly...
>>>>
>>>> Whenever I try to register the user alice, I get the following error
>>>> from Asterisk:
>>>>
>>>> [Apr 13 19:45:47] ERROR[1706]: chan_sip.c:14798 register_verify: Peer
>>>> 'alice' is trying to register, but not configured as host=dynamic
>>>> [Apr 13 19:45:47] NOTICE[1706]: chan_sip.c:24905
>>>> handle_request_register: Registration from '"alice"
>>>> <sip:alice at swannsips.com>' failed for '192.168.5.10:5060' - Peer is not
>>>> supposed to register
>>>>
>>>> From looking at the Asterisk documentation I see that this *seems* to
>>>> be correct since we do not declare the new users with "host=dynamic" in
>>>> the database.
>>>>
>>>> Is this correct and the fact that the tutorial tells one to "register"
>>>> the user an error? Or is there clearly something else I am missing?
>>>>
>>>> Any help on this would be greatly appreciated since I've spent the
>>>> better part of the last week trying to resolve this, including trying to
>>>> change the database layout itself to support "host=dynamic" - see my
>>>> post on the asterisk forum here:
>>>> http://forums.asterisk.org/viewtopic.php?f=1&t=86342 although that
>>>> didn't really help much...
>>>>
>>>> Kind regards,
>>>>
>>>> Steven
>>>>
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