Thank you very much Sammy that worked
Ian
On 29 January 2013 18:14, <sr-users-request(a)lists.sip-router.org> wrote:
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> 1. Re: Asterisk and dispatcher (Ian French)
> 2. Re: Asterisk and dispatcher (SamyGo)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 29 Jan 2013 17:55:40 +0000
> From: Ian French <fretec(a)gmail.com>
> Subject: Re: [SR-Users] Asterisk and dispatcher
> To: sr-users(a)lists.sip-router.org
> Message-ID:
> <CAOAGGZSr_24VDL8508XJo9LjkvR=
> uo67JoM_MPPyd3bVbWxC7Q(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote:
>
> > Send sr-users mailing list submissions to
> > sr-users(a)lists.sip-router.org
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> > or, via email, send a message with subject or body 'help' to
> > sr-users-request(a)lists.sip-router.org
> >
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> > than "Re: Contents of sr-users digest..."
> >
> >
> > 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(a)gmail.com>
> > Subject: Re: [SR-Users] Asterisk and dispatcher
> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> > Users Mailing List" <sr-users(a)lists.sip-router.org>
> > Message-ID:
> > <
> > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)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(a)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(a)lists.sip-router.org
> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> > >
> > >
> >
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Content preview: Hi I have been using your excellent Voip facility for sometime.
Recently I have encountered that when I called from 'phone A' to 'Phone B'
connects and can hear Phone B's voice but can't hear phone A's voice when
dialed from phone A. But when Phone B made the call to phone A, can hear
both parties voices. This problem started very recently. I haven't done any
configuration changes. This problem incurred nearly a week or two ago. I
experienced same problem long time ago and it resolved after few days time.
But this time it didn't resolved. My SIP no 87415661. I would appreciate
if you could kindly help me to rectify the problem. [...]
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Hi Andrew,
Here is one call with special characters:
call-id = 100c8a98-c0a8016b-13c4-50029-203fa-68e78324-203fa
How we can block such attempts, by rewrite rules?
Regards
Ivan
Hi I am trying to build the kamailio deb packages on ubuntu 12.04.
when I run make deb I get
dpkg-checkbuilddeps: Unmet build dependencies: libmono-dev
but libmono-dev is not available on 12.04 how do I
1. Fix the dependancy
2. Ignore it as I am not using app_mono (although I have an idea that may
in the future)
Thanks
Gareth
