[SR-Users] Asterisk and dispatcher
Ian French
fretec at gmail.com
Tue Jan 29 18:55:40 CET 2013
On 24 January 2013 08:16, <sr-users-request at lists.sip-router.org> wrote:
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> Today's Topics:
>
> 1. Re: Asterisk and dispatcher (SamyGo)
> 2. Re: Asterisk and dispatcher (SamyGo)
> 3. Kamailio Multi tenant as registrar server ony (andrea mucci)
> 4. help me about siremis. (???)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 23 Jan 2013 22:26:26 -0800
> From: SamyGo <govoiper at gmail.com>
> Subject: Re: [SR-Users] Asterisk and dispatcher
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
> <
> CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> You've metioned this already in your code:
>
> if(ds_is_from_list("1"))
>
> Is this not working for you?
>
> Regards,
> Sammy
>
>
>
> On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec at gmail.com> wrote:
>
> > Hi,
> >> I've been working my way through this tutorial (
> >>
> http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
> )
> >> and all it working well thanks. In the other benifits section you
> mention
> >> that this configuration can be used with multiple instances of Asterisk.
> >> I've configured my [TOASTERISK] route
> >
> >
> > route[TOASTERISK] {
> > ds_select_dst("1","4");
> > $sht(forw=>$ft)=$du;
> > sl_send_reply("100","Trying");
> > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk
> > Cluster [$rm:$au].\n");
> > route(RELAY);
> > exit;
> >
> > How can i achieve the same for my [FROMASTERISK] route as now with
> >
> > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address"
> > asterisk.bindport = "5060" desc "Asterisk Port"
> >
> > and
> >
> > route[FROMASTERISK] {
> > if($si==$sel(cfg_get.asterisk.bindip)
> > && $sp==$sel(cfg_get.asterisk.bindport))
> > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk
> > Cluster To [$rm:$au].\n");
> > return 1;
> > return -1;
> > }
> >
> > This allows calls only from cfg_get.asterisk.bindip (single gateway)
> where
> > I would like gateways in the dispatcher list to be selected and marked
> as a
> > positive match
> >
> > if(ds_is_from_list("1"))
> > $sht(forw=>$ft)=$si;
> > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk
> > Cluster To [$rm:$au].\n");
> > return 1;
> > return -1;
> > }
> >
> > Can you help?
> >
> > Thanks in advance
> > Ian
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
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>
> ------------------------------
> Thanks Sammy but I am confused what curly brackets am i missing
>
if(ds_is_from_list("1"))
$sht(forw=>$ft)=$si;
xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To
[$rm:$au].\n");
return 1;
return -1;
Thks in advance
Ian
> Message: 2
> Date: Wed, 23 Jan 2013 22:29:25 -0800
> From: SamyGo <govoiper at gmail.com>
> Subject: Re: [SR-Users] Asterisk and dispatcher
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
> <
> CAJUJwtgMna+5S5WF29pn__xpb-NQoSyCCYibZyRRGGT2K0r3Ng at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Except that I find missing curly brackets for the if() condition which will
> always return 1/positive.
>
>
> On Wed, Jan 23, 2013 at 10:26 PM, SamyGo <govoiper at gmail.com> wrote:
>
> > Hi,
> >
> > You've metioned this already in your code:
> >
> > if(ds_is_from_list("1"))
> >
> > Is this not working for you?
> >
> > Regards,
> > Sammy
> >
> >
> >
> > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec at gmail.com> wrote:
> >
> >> Hi,
> >>> I've been working my way through this tutorial (
> >>>
> http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
> )
> >>> and all it working well thanks. In the other benifits section you
> mention
> >>> that this configuration can be used with multiple instances of
> Asterisk.
> >>> I've configured my [TOASTERISK] route
> >>
> >>
> >> route[TOASTERISK] {
> >> ds_select_dst("1","4");
> >> $sht(forw=>$ft)=$du;
> >> sl_send_reply("100","Trying");
> >> xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk
> >> Cluster [$rm:$au].\n");
> >> route(RELAY);
> >> exit;
> >>
> >> How can i achieve the same for my [FROMASTERISK] route as now with
> >>
> >> asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address"
> >> asterisk.bindport = "5060" desc "Asterisk Port"
> >>
> >> and
> >>
> >> route[FROMASTERISK] {
> >> if($si==$sel(cfg_get.asterisk.bindip)
> >> && $sp==$sel(cfg_get.asterisk.bindport))
> >> xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk
> >> Cluster To [$rm:$au].\n");
> >> return 1;
> >> return -1;
> >> }
> >>
> >> This allows calls only from cfg_get.asterisk.bindip (single gateway)
> >> where I would like gateways in the dispatcher list to be selected and
> >> marked as a positive match
> >>
> >> if(ds_is_from_list("1"))
> >> $sht(forw=>$ft)=$si;
> >> xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk
> >> Cluster To [$rm:$au].\n");
> >> return 1;
> >> return -1;
> >> }
> >>
> >> Can you help?
> >>
> >> Thanks in advance
> >> Ian
> >> _______________________________________________
> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> >> sr-users at lists.sip-router.org
> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >>
> >>
> >
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> ------------------------------
>
> Message: 3
> Date: Wed, 23 Jan 2013 22:37:54 +0100
> From: andrea mucci <amucci at hotmail.es>
> Subject: [SR-Users] Kamailio Multi tenant as registrar server ony
> To: sr-users at lists.sip-router.org
> Message-ID: <BLU0-SMTP39596AF0F6043976A9F3860A8150 at phx.gbl>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Gurus,
>
> i have a basic question.
>
> i have a SIP PBX server that don't have the Multi Tenant Feature or Multi
> Company Feature.
> i now that Kamailio support this feature, so
>
> is possible to use the kamailio as registrar server and redirect all the
> services, SUBSCRIBE, NOTIFY, and RTP to the SIP PBX with corrects headers?
>
> Thanks for your help
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 24 Jan 2013 14:11:43 +0900 (KST)
> From: ??? <whdals0 at naver.com>
> Subject: [SR-Users] help me about siremis.
> To: sr-users at lists.sip-router.org
> Message-ID: <47ea2c39c02e2ff4796fa2165b853b45 at tweb10.nm.nhnsystem.com>
> Content-Type: text/plain; charset="utf-8"
>
> I'm developers who trying to install siremis.
>
> will go in there and install siremis website.
>
> However, server-services -> domain list by pressing will get the
> following error.
> ?
> ?
>
>
> What can I do to fix this?
> ?
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