On 24 January 2013 08:16, <sr-users-request@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@gmail.com>
Subject: Re: [SR-Users] Asterisk and dispatcher
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
        Users   Mailing List" <sr-users@lists.sip-router.org>
Message-ID:
        <CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ@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@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@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@gmail.com>
Subject: Re: [SR-Users] Asterisk and dispatcher
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
        Users   Mailing List" <sr-users@lists.sip-router.org>
Message-ID:
        <CAJUJwtgMna+5S5WF29pn__xpb-NQoSyCCYibZyRRGGT2K0r3Ng@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@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@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@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@hotmail.es>
Subject: [SR-Users] Kamailio Multi tenant as registrar server ony
To: sr-users@lists.sip-router.org
Message-ID: <BLU0-SMTP39596AF0F6043976A9F3860A8150@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@naver.com>
Subject: [SR-Users] help me about siremis.
To: sr-users@lists.sip-router.org
Message-ID: <47ea2c39c02e2ff4796fa2165b853b45@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 -&gt; domain list by pressing will get the following error.
?
?


What can I do to fix this?
?
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