[OpenSIPS-Users] - Not sending ACK back!

Sat Oct 22 22:03:24 UTC 2022

Hi Nitesh

I think the default OPENSIPS.CFG script has a Registrar Server and the
Lookup function. Please take a look there. To call then, you have to use
lookup function.

The ACK missing feature maybe related to NAT problems. Where is the ACK
lost? Did origin sent to Opensips? Did opensips forward to destination?
Please take a deep look at default opensips.cfg, I tested it in the past
and worked fine.

On Fri, Oct 21, 2022 at 3:05 PM Nitesh Divecha <aviator.nitesh.d at gmail.com>
wrote:

> Daniel,
>
> Thanks for your detailed email. Much appreciated!
>
> Yes I totally understand about OpenSIPS and I'm going over all the
> documentations and tutorials I can find! My background is from Asterisk, so
> with that mentality I'm tackling OpenSIPS.
>
> Asterisk (Context) vs OpenSIPS (C-style script), it is a huge challenge
> but getting the hang of it!
>
> Lets pick option (3) Routing local calls (ATA to ATA)... Do you have any
> examples I can follow and set up a local route? I set up two extensions
> (8883456 and 7773456) via OpenSIPS Control Panel and I was able to register
> two ATA's. I do have the Registrar module loaded... How can I call each
> other?
>
> #### REGISTRAR module
>
> loadmodule "registrar.so"
>
> modparam("registrar", "default_expires", 3600)
>
> modparam("registrar", "min_expires", 60)
>
> modparam("registrar", "max_expires", 120)
>
> # modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
>
> # modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment
> the next line not to allow more than 10 contacts per AOR */
>
> modparam("registrar", "max_contacts", 10)
>
> modparam("registrar", "received_avp", "$avp(rcv)")
>
> modparam("registrar", "retry_after", 30)
>
>
> Regarding option (4) - I have both options. IP to IP and User/Pass
> authentication provider. Using User/Pass I was able to register OpenSIPS as
> UAC to a remote server and I was able to make outbound calls but call keeps
> on dropping due to no ACK.
>
> Cheers,
> Nitesh
>
>
>
>
>
> On Fri, Oct 21, 2022 at 1:26 PM Daniel Zanutti <daniel.zanutti at gmail.com>
> wrote:
>
>> Hi Nitesh
>>
>> As you already know, opensips is a low level software. You have to
>> understand several aspects of SIP, network, RTP, DNS that when you use
>> Asterisk, most you don't need to understand deep.
>>
>> Trying to help you, your script is way simple for you achievements. You
>> need:
>> 1) Check NAT on all request + all replies. This is to fix your SIP
>> messages.
>>
>> 2) Check if you need to apply RTPPROXY on the call. You can use the
>> "engage" function on INVITE then forget about it OR you can use manual way
>> with "offer" function and handle all scenarios manually. For example, call
>> the "answer" function on the 200 OK. Then delete on BYE.
>>
>> 3) Routing local calls (ATA to ATA) you need to handle the Register first
>> with "save" function, then later handle the INVITE with the "lookup"
>> function, both of Registrar module.
>>
>> 4) PSTN can be used as a direct route or some dynamic routing solution.
>> Make it work first with direct routing. Need to check how authenticate
>> works on your carrier. If IP based will be fine, if user/pass you need to
>> make your opensips authenticate, it's a little harder.
>>
>> 5) DID - You have to create some specific INBOUND rules. Calls will be
>> anonymous or authenticated?
>>
>> 6) Fax - Better solve other issues first.
>>
>> Hope this gives you some direction. Look for some tutorials.
>>
>> Regards
>>
>> On Fri, Oct 21, 2022 at 11:11 AM Nitesh Divecha <
>> aviator.nitesh.d at gmail.com> wrote:
>>
>>> Hello All,
>>>
>>> I have been scratching my head for a few days now... Just to recap:
>>>
>>> I'm a newbie with OpenSIPS so bear with me... I got OpenSIPS 3.3.1
>>> (residential) running on Debian 11 with OpenSIPS Control Panel 9.3.2 and
>>> MySQL.
>>>
>>> My goal is to:
>>> 1) Make two ATA's register and call each other (locally)... *Stopped
>>> working, I think routing logic is missing.*
>>> 2) Make ATA to call PSTN via an outbound SIP trunk or DID provider... *No
>>> ACK sent to Outbound provider.*
>>> 3) Receive inbound calls from PSTN or SIP trunk and forward it to
>>> registered ATA... *Getting rejected.*
>>> 4) Able to send and receive faxes from and to PSTN... *Haven't even
>>> touched.*
>>>
>>> Fast forward... I did achieve a few of my goals but they stopped
>>> working... You fix one thing and you break others...
>>>
>>> My current issue is OpenSIPS is not sending ACK back to the Outbound
>>> provider when I make calls to PSTN thus calls are getting dropped from the
>>> Outbound provider due to no ACK. This issue started when I implemented
>>> topology_hiding('C"), rtpproxy_offer("ro"), uac_replace_from( ,
>>> "$avp(furi)").
>>>
>>> Here is my code snippet:
>>>
>>> ####### Routing Logic ########
>>>
>>>
>>> # main request routing logic
>>>
>>>
>>> route{
>>>
>>>
>>> #if ($rU=~"^\+[1-9][0-9]+$") {
>>>
>>>         if (dp_translate(10 ,$rU ,$rU) ) {
>>>
>>>                 xlog("*** 2. Dial plan translate from source $avp(src)
>>> to $rU ***\n");
>>>
>>>
>>>                 $avp(furi) = "sip:aaabbbcccc at gothamcity.com";
>>>
>>>                 uac_replace_from( , "$avp(furi)");
>>>
>>>                 #strip(1);
>>>
>>>                 if (!do_routing(0)) {
>>>
>>>                         send_reply(500,"No PSTN Route found");
>>>
>>>                         exit;
>>>
>>>                 }
>>>
>>>                 # t_on_branch("change_from");
>>>
>>>                 route(relay);
>>>
>>>                 exit;
>>>
>>>         }
>>>
>>>
>>>
>>> route[relay] {
>>>
>>>         xlog("*** 3. Entering route relay ***\n");
>>>
>>>         # for INVITEs enable some additional helper routes
>>>
>>>         if (is_method("INVITE")) {
>>>
>>>                 topology_hiding("C");
>>>
>>>                 if(remove_hf("User-Agent")){
>>>
>>>                         xlog("*** 4. User-Agent found and removed.
>>> ***\n");
>>>
>>>                 }
>>>
>>>
>>>                 if (isflagset("NAT") && has_body("application/sdp")) {
>>>
>>>                         rtpproxy_offer("ro");
>>>
>>>                         #rtpproxy_offer();
>>>
>>>                 }
>>>
>>>
>>>                 t_on_branch("per_branch_ops");
>>>
>>>                 t_on_reply("handle_nat");
>>>
>>>                 t_on_failure("missed_call");
>>>
>>>         }
>>>
>>>
>>>         if (isflagset("NAT")) {
>>>
>>>                 add_rr_param(";nat=yes");
>>>
>>>         }
>>>
>>>
>>>         if (!t_relay()) {
>>>
>>>                 send_reply(500,"Internal Error");
>>>
>>>         }
>>>
>>>         exit;
>>>
>>> }
>>>
>>> Any thoughts or suggestions on what to check for ACK?
>>>
>>> Cheers,
>>> Nitesh
>>>
>>>
>>>
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>>> Users at lists.opensips.org
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>>>
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