[OpenSIPS-Users] Database entries for Trunking script?
jack krog
jackkrog at yahoo.com
Wed Oct 21 00:14:26 EST 2020
Yup, It seems to match?
Oct 20 23:56:07 [795] [773bd43a4da52bc067b98c57352d2f93 at PBX.net] Request username is [2111234567]
On Tuesday, October 20, 2020, 06:48:44 AM MST, Răzvan Crainea <razvan at opensips.org> wrote:
OK, then the cache is OK. Then I guess you're probably modifying the
request username just before the do_routing in your code.
Can you add an xlog just above your do_routing line, to print the
Request username? You should use something like:
xlog("[$ci] Request username is [$rU]\n");
Best regards,
Răzvan
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 10/16/20 6:21 PM, jack krog via Users wrote:
>
> mi dr_number_routing group_id=1 number=2111234567
> {
> "Matched Prefix": "2111234567",
> "GATEWAY": "cust0001",
> "CARRIER": "cust0001"
> }
>
> I had a missing '1' in my number below, I'm using 2111234567 everywhere.
> On Thursday, October 15, 2020, 02:49:32 AM MST, Răzvan Crainea
> <razvan at opensips.org> wrote:
>
>
> What does dr_number_routing MI command [1] for group 1 and the number
> you're using output?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 10/14/20 9:10 PM, jack krog via Users wrote:
> > Razvan, Thank you for your response,
> >
> > I have a gateway setup for both the incoming host (pbxb) and my PBX
> > (cust0001)
> >
> > https://imgur.com/ZXjitEd
> >
> > I have rules set up, specifically one for 211234567
> >
> > https://imgur.com/YR7YucB
> >
> > I even tried adding a group incase I misunderstood you.
> >
> > https://imgur.com/E6xZngY
> >
> > and I'm still failing at this step:
> >
> > # route calls based on prefix
> > if ( !do_routing(1) ) {
> > send_reply(404,"No Route found");
> > exit;
> > }
> >
> > How can I dig into what's going on here, I tried starting opensips in
> > Debug mode but the output is quite verbose and doesn't mean anything to
> > me yet.
> >
> > Additionally, is there a easier way to reload rules than to kill
> > opensips and restart it? I saw reference to a opensipsctl command but my
> > install doesn't have it.
> > On Wednesday, October 14, 2020, 08:20:12 AM MST, Răzvan Crainea
> > <razvan at opensips.org <mailto:razvan at opensips.org>> wrote:
> >
> >
> > Hi, Jack!
> >
> > Rules have to be added in group 1[1].
> > You need to add a gateway to host 172.18.0.8. Then you need to add a
> > rule with the prefix (2111234567 in your case) and assign it to group 1.
> > If you don't have group 1, you can provision it in the settings tab.
> >
> > [1]
> >
> https://github.com/OpenSIPS/opensips/blob/master/menuconfig/configs/opensips_trunking.m4#L268
> >
> > Best regards,
> >
> > Răzvan Crainea
> > OpenSIPS Core Developer
> > http://www.opensips-solutions.com
> >
> > On 10/12/20 7:18 AM, jack krog via Users wrote:
> > > Bump?
> > >
> > > I still don't understand how drouting needs to be configured for the
> > > trunking script to work.
> > >
> > > Are there videos covering the scripts generated by osipsconfig?
> > > On Thursday, October 1, 2020, 10:02:50 AM MST, jack krog via Users
> > > <users at lists.opensips.org <mailto:users at lists.opensips.org>
> <mailto:users at lists.opensips.org <mailto:users at lists.opensips.org>>> wrote:
> > >
> > >
> > > Hello,
> > >
> > > I'm new to opensips coming from a asterisk background.
> > >
> > > I'm trying to learn how things work by using the osipsconfig generated
> > > trunking script but I can't find documentation on what database
> values I
> > > need to input to route a call to my PBX.
> > >
> > > The script appears to get to where it runs dp_translate followed by
> > > do_routing and then all my test calls get rejected with a No Route
> Found
> > > response.
> > >
> > > /What do I need to put in the control panel to route the DDI number
> > > 211-123-4567 to the host 172.18.0.8 using the domain cust0001.pbx ? /
> > >
> > > I've entered my PBX under Dynamic Routing > Gateways but don't
> know if I
> > > need to create a Group?
> > >
> > > If I want to pass the number through without changing it do I need to
> > > use dialplan? Is that where I need to point my DID to my PBX?
> > >
> > > Thank you for your time.
> > > _______________________________________________
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> > >
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> >
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