[Freeswitch-users] Opensips integration questions

[Schneur Rosenberg]

Thanks, set_user is exactly what I was looking for, I will test it tomorrow.

On Oct 11, 2016 01:22, "Abaci B" <abaci64 at gmail.com> wrote:

> see https://freeswitch.org/jira/browse/FS-5009
> or you can simply use the set_user https://freeswitch.org/
> confluence/display/FREESWITCH/mod_dptools%3A+set+user application to do
> it
> with either option be extremely careful to ensure it's only applied to
> calls coming from proxy and that the proxy already authed the user
> (otherwise anyone would be able to use credentials belonging to someone
> else)
>
> On Thu, Sep 29, 2016 at 6:09 AM, Schneur Rosenberg <thesipguy at gmail.com>
> wrote:
>
>> **UPDATE**
>>
>> I looked into my sip traces and found the NAT problem, Freeswitch sends a
>> UPDATE request to set the P-Asserted-Identity to "Outbound Call", the
>> problem was my OpenSIPS was not set to t_on_reply to go to my reply route
>> that fixes nat, so it never did the  fix_nated_contact() therefore
>> all sequential requests used the bad contact from the reply of the update,
>> on the other hand my asterisk box never sent that UPDATE therefore is used
>> the "contact" from the initial INVITE which was fixed by OpenSIPS, I
>> configured OpenSIPS to process the correct reply route for the UPDATE and
>> problem was solved.
>>
>> I'm still looking for a answer to my first question.
>>
>> thank you
>> S. Rosenberg
>>
>> On Wed, Sep 28, 2016 at 1:32 PM, Schneur Rosenberg <thesipguy at gmail.com>
>> wrote:
>>
>>> Hi, I'm pretty new to Freeswitch, I built a complete solution using
>>> OpenSIPS/Asterisk and I'm in the process of replacing Asterisk with
>>> Freeswitch.
>>>
>>> My current system is built that OpenSIPS act as a registrar and as a
>>> load balancer, the rest is done on a cluster of Asterisk machines, which as
>>> I said before is being rewritten to Freeswitch, in Freeswitch I'm using
>>> xml_curl for my dialplan and call routing, I'm using Intralanman project to
>>> handle the sip users.
>>>
>>> With Asterisk I had OpensSIPS as a trunk and I know that everything
>>> coming from that trunk, is either a DID or a authenticated outgoing call,
>>> Asterisk and OpenSIPS shared the user table, Asterisk recognized the sip
>>> user name from the FROM field and used all the variables etc from its own
>>> sip user table as if the user has registered directly to itself, I had some
>>> issues with blind transfers but I was able to do some manipulations and DB
>>> queries to handle it fine.
>>>
>>> I tried doing the same with Freeswitch, I let the ACL allow the ip from
>>> Freeswitch, but Freeswitch would not use the variables from the user like
>>> Asterisk did, for example accountcode, user_context etc, I was able to
>>> partially fix that by having my web server do a db query and then using the
>>> "set" application set all the variables, but I'm not sure if its the proper
>>> solution, does anyone have a better solution? will setting my OpenSIPS as a
>>> gateway instead of merely allowing it in with ACL work better? right now
>>> I'm using the application "execute_extension" to send it to the proper
>>> context that I've pulled from the DB, I dont like the hack and if there is
>>> a better solution I would like to know.
>>>
>>> One more major problem that I have is NAT related, when I place a
>>> outgoing call for example from a phone which is behind a NAT (all of our
>>> customers phones are behind NAT) the call connects and audio works 2 ways,
>>> but when the called party hangs up, the carrier sends the BYE to
>>> Freeswitch, freeswitch then sends it to my OpenSIPS which is supposed to
>>> send it to my clients phone, but Freeswich writes the RURI with the natted
>>> address, the RURI is BYE sip:101 at 192.168.0.11:5062 SIP/2.0. and my
>>> OpenSIPS does not know what to do with it and ignores it, my Asterisk
>>> replaced the RURI with the received address in the VIA, so same RURI would
>>> be BYE sip:101 at 64.69.47.175:5062 SIP/2.0. which worked fine, same
>>> problem happens with all subsequent requests  if user does not hang up, I
>>> need a solution for this major issue.
>>>
>>> thank you
>>> S. Rosenberg
>>>
>>>
>>
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>
>
> _________________________________________________________________________
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>
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>
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