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Wed Dec 21 17:12:16 CET 2011
On Mon, Feb 6, 2012 at 8:13 PM, Greg Mannie <greg at latigi.com> wrote:
> Thank you again for your help.
>
> I have been reading on the dispatcher module and have a question. It
> would seem many people use Kamailio as more than proxy and register sip
> extensions against it. Since we host virtual pbx (asterisk 1.8, Freepbx)
> for a few different clients, each instance is separate with it's own
> database.
>
> I am having problems wrapping my head around the configuration I should
> use. Is there not a method just to add DID in the same fashion as asterisk.
> So I register a trunk on Kamailio and based on incoming DID it sends it to
> the correct asterisk server?
>
> I have my asterisk 1.8 on a public ip address with a trunk registered to
> the Kamailio server which also has a public ip address.
>
>
> Regards,
>
> Greg
>
>
>
>
> Quoting Stoyan Mihaylov <stoyan.v.mihaylov at gmail.com>:
>
> I dont know about siremis, but you can forward calls to different groups
>> of
>> Asterisk servers - using
>> ds_select_dst(set, alg);
>> Where set is set of Asterisk servers - you can check that module.
>> The problem is - I have no idea how you can select different sets in
>> kamailio.cfg, except by length, or some matching pattern in CallerID.
>> But if you put whole logic used in Asterisk in DB - then you dont care
>> which server will take the call, because you can put whole logic purely in
>> DB - including extensions etc.
>> At least I prefer to have almost nothing in extensions.conf - and
>> everything to stay either in DB or in AGI scripts.
>>
>> My knowledge of Kamailio is very very basic - I know only few things
>> there.
>> Asterisk and Kamailio can run on same server, but I cant see any reason
>> for
>> that. I mean you will have lot of troubles in such case, and nothing
>> "good". This is only if you want to make some tests. But you can expect
>> lot
>> of troubles.
>>
>>
>> On Fri, Feb 3, 2012 at 9:40 PM, Greg Mannie <greg at latigi.com> wrote:
>>
>> Thank you for your detailed response. Sorry for the trouble but would
>>> you
>>> be able to also answer the following.
>>>
>>> Do you know if this same type of deployment would be suited to our needs.
>>> Many of the Asterisk servers we host are for clients, who have their own
>>> extensions, voicemail, ivr etc. I was hoping I could setup routes on the
>>> kamailio and direct them to the appropriate asterisk server.
>>>
>>> Initially I thought it would be as simple as setting up an inbound route
>>> on Asterisk. Ha.. I also installed siremis 3.2 and perhaps reading on
>>> how
>>> to use it will provide clearer details.
>>>
>>> I know so little, I'm not even sure if I need to have Kamailio and
>>> Asterisk running on the same server, since I only want Kamailio as a
>>> proxy.
>>>
>>>
>>> Regards,
>>>
>>> Greg
>>>
>>>
>>> Quoting Stoyan Mihaylov <stoyan.v.mihaylov at gmail.com>:
>>>
>>> We were in similar situation. Many years with Asterisk and then we were
>>>
>>>> forced to use ser - and we preferred Kamailio.
>>>> Now we do:
>>>> Kamailio has global IP address and clients register to it.
>>>> Kamailio forward all calls to Asterisk boxes using following:
>>>> ds_select_dst("1","4");#You can use many asterisk boxes this way
>>>> $sht(forw=>$ft)=$du; #this way I store used path
>>>> I used t_relay, instead of forward, because my Asterisks are with local
>>>> IP.
>>>> Calls from Asterisk are send to Kamailio if they are to local user, or
>>>> to
>>>> our SIP provider. There are no problems with calls from Asterisk to SIP
>>>> provider, even if Asterisk is behind NAT.
>>>> Asterisk accepts calls from SIP provider though registrar lines in
>>>> sip.conf. Asterisk can forward calls from our SIP provider to local
>>>> users
>>>> in Kamailio.
>>>> I got problems with ACK and BYE. To solve them, I used
>>>> if(($td=="sip.name.of.**kamail**io.server.com<http://kamailio.server.com>
>>>> <http://sip.name.**of.kamailio.server.com<http://sip.name.of.kamailio.server.com>
>>>> >
>>>> ")||($si=="**IPofServer")){
>>>>
>>>> $du=$sht(forw=>$ft);
>>>> }
>>>>
>>>> On Fri, Feb 3, 2012 at 8:13 PM, Greg Mannie <greg at latigi.com> wrote:
>>>>
>>>> Hello
>>>>
>>>>>
>>>>> After much reading I have come to the realization that after years of
>>>>> using Asterisk I know very little about Sip.
>>>>>
>>>>> I have my Kamailio box up, I have Asterisk 1.8.x running with realtime
>>>>> working. I thought it would be just a case of registering SIP trunks
>>>>> from
>>>>> my provider to the kamailio and registering our internal asterisk
>>>>> servers
>>>>> to the kamailio.
>>>>>
>>>>> Much of what I read talks about using Asterisk as the PSTN interface,
>>>>> but
>>>>> that interface is through a sip trunk purchased from a provider. Won't
>>>>> Kamailio be the PSTN gateway? The idea here is to pool all the sip
>>>>> trunks
>>>>> from the various hosted asterisk solutions (VM running asterisk) and
>>>>> point
>>>>> them all to a proxy to facilitate the aggregation of traffic.
>>>>>
>>>>> I have been reading SIP tutorials and the mailing list archives. If
>>>>> anyone has a sample config and perhaps a little direction it would be
>>>>> highly appreciated.
>>>>>
>>>>> Thank you
>>>>>
>>>>> Greg
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ______________________________******_________________
>>>>>
>>>>>
>>>>> 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<http://lists.sip-router.org/****cgi-bin/mailman/listinfo/sr-****users>
>>>>> <http://lists.sip-**router.org/**cgi-bin/mailman/**listinfo/sr-**users<http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users>
>>>>> >
>>>>> <http://lists.sip-router.****org/cgi-bin/mailman/listinfo/***
>>>>> *sr-users<http://lists.sip-**router.org/cgi-bin/mailman/**
>>>>> listinfo/sr-users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>
>>>
>>> ______________________________****_________________
>>> 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<http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users>
>>> <http://lists.sip-router.**org/cgi-bin/mailman/listinfo/**sr-users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>> >
>>>
>>>
>>
>
>
> ______________________________**_________________
> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
--14dae9340e2beaa60e04b85070db
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I suppose this is better for you:<div><a href=3D"http://kamailio.org/docs/m=
odules/3.2.x/modules/carrierroute.html#id2548186">http://kamailio.org/docs/=
modules/3.2.x/modules/carrierroute.html#id2548186</a></div><div>But I never=
used it - we just did not need anything alike.=A0</div>
<div>From overview - you can define routing tree per user.<br><br><div clas=
s=3D"gmail_quote">On Mon, Feb 6, 2012 at 8:13 PM, Greg Mannie <span dir=3D"=
ltr"><<a href=3D"mailto:greg at latigi.com">greg at latigi.com</a>></span> =
wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Thank you again for your help.<br>
<br>
I have been reading on the dispatcher module and have a question. =A0It wou=
ld seem many people use Kamailio as more than proxy and register sip extens=
ions against it. =A0Since we host virtual pbx (asterisk 1.8, Freepbx) for a=
few different clients, each instance is separate with it's own databas=
e.<br>
<br>
I am having problems wrapping my head around the configuration I should use=
. Is there not a method just to add DID in the same fashion as asterisk. =
=A0So I register a trunk on Kamailio and based on incoming DID it sends it =
to the correct asterisk server?<br>
<br>
I have my asterisk 1.8 on a public ip address with a trunk registered to th=
e Kamailio server which also has a public ip address.<div><div class=3D"h5"=
><br>
<br>
Regards,<br>
<br>
Greg<br>
<br>
<br>
<br>
<br>
Quoting Stoyan Mihaylov <<a href=3D"mailto:stoyan.v.mihaylov at gmail.com" =
target=3D"_blank">stoyan.v.mihaylov at gmail.com</a>>:<br>
<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5">
I dont know about siremis, but you can forward calls to different groups of=
<br>
Asterisk servers - using<br>
ds_select_dst(set, alg);<br>
Where set is set of Asterisk servers - you can check that module.<br>
The problem is - I have no idea how you can select different sets in<br>
kamailio.cfg, except by length, or some matching pattern in CallerID.<br>
But if you put whole logic used in Asterisk in DB - then you dont care<br>
which server will take the call, because you can put whole logic purely in<=
br>
DB - including extensions etc.<br>
At least I prefer to have almost nothing in extensions.conf - and<br>
everything to stay either in DB or in AGI scripts.<br>
<br>
My knowledge of Kamailio is very very basic - I know only few things there.=
<br>
Asterisk and Kamailio can run on same server, but I cant see any reason for=
<br>
that. I mean you will have lot of troubles in such case, and nothing<br>
"good". This is only if you want to make some tests. But you can =
expect lot<br>
of troubles.<br>
<br>
<br>
On Fri, Feb 3, 2012 at 9:40 PM, Greg Mannie <<a href=3D"mailto:greg at lati=
gi.com" target=3D"_blank">greg at latigi.com</a>> wrote:<br>
<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5">
Thank you for your detailed response. =A0Sorry for the trouble but would yo=
u<br>
be able to also answer the following.<br>
<br>
Do you know if this same type of deployment would be suited to our needs.<b=
r>
=A0Many of the Asterisk servers we host are for clients, who have their own=
<br>
extensions, voicemail, ivr etc. =A0I was hoping I could setup routes on the=
<br>
kamailio and direct them to the appropriate asterisk server.<br>
<br>
Initially I thought it would be as simple as setting up an inbound route<br=
>
on Asterisk. Ha.. =A0I also installed siremis 3.2 and perhaps reading on ho=
w<br>
to use it will provide clearer details.<br>
<br>
I know so little, I'm not even sure if I need to have Kamailio and<br>
Asterisk running on the same server, since I only want Kamailio as a proxy.=
<br>
<br>
<br>
Regards,<br>
<br>
Greg<br>
<br>
<br>
Quoting Stoyan Mihaylov <<a href=3D"mailto:stoyan.v.mihaylov at gmail.com" =
target=3D"_blank">stoyan.v.mihaylov at gmail.com</a>>:<br>
<br>
=A0We were in similar situation. Many years with Asterisk and then we were<=
br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5">
forced to use ser - and we preferred Kamailio.<br>
Now we do:<br>
Kamailio has global IP address and clients register to it.<br>
Kamailio forward all calls to Asterisk boxes using following:<br>
=A0ds_select_dst("1","4");#You can use many asterisk bo=
xes this way<br>
=A0$sht(forw=3D>$ft)=3D$du; #this way I store used path<br>
I used t_relay, instead of forward, because my Asterisks are with local<br>
IP.<br>
Calls from Asterisk are send to Kamailio if they are to local user, or to<b=
r>
our SIP provider. There are no problems with calls from Asterisk to SIP<br>
provider, even if Asterisk is behind NAT.<br>
Asterisk accepts calls from SIP provider though registrar lines in<br>
sip.conf. Asterisk can forward calls from our SIP provider to =A0local user=
s<br>
in Kamailio.<br>
I got problems with ACK and BYE. To solve them, I used<br></div></div>
if(($td=3D=3D"sip.name.of.**<a href=3D"http://kamailio.server.com" tar=
get=3D"_blank">kamail<u></u>io.server.com</a><<a href=3D"http://sip.name=
.of.kamailio.server.com" target=3D"_blank">http://sip.name.<u></u>of.kamail=
io.server.com</a>><br>
")||($si=3D=3D"**IPofServer")){<div class=3D"im"><br>
=A0$du=3D$sht(forw=3D>$ft);<br>
}<br>
<br>
On Fri, Feb 3, 2012 at 8:13 PM, Greg Mannie <<a href=3D"mailto:greg at lati=
gi.com" target=3D"_blank">greg at latigi.com</a>> wrote:<br>
<br>
=A0Hello<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
<br>
After much reading I have come to the realization that after years of<br>
using Asterisk I know very little about Sip.<br>
<br>
I have my Kamailio box up, I have Asterisk 1.8.x running with realtime<br>
working. =A0I thought it would be just a case of registering SIP trunks<br>
from<br>
my provider to the kamailio and registering our internal asterisk servers<b=
r>
to the kamailio.<br>
<br>
Much of what I read talks about using Asterisk as the PSTN interface, but<b=
r>
that interface is through a sip trunk purchased from a provider. =A0Won'=
;t<br>
Kamailio be the PSTN gateway? =A0The idea here is to pool all the sip<br>
trunks<br>
from the various hosted asterisk solutions (VM running asterisk) and<br>
point<br>
them all to a proxy to facilitate the aggregation of traffic.<br>
<br>
I have been reading SIP tutorials and the mailing list archives. =A0If<br>
anyone has a sample config and perhaps a little direction it would be<br>
highly appreciated.<br>
<br>
Thank you<br>
<br>
Greg<br>
<br>
<br>
<br>
<br>
<br></div>
______________________________<u></u>****_________________<div class=3D"im"=
><br>
<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href=3D"mailto:sr-users at lists.sip-router.org" target=3D"_blank">sr-users=
@lists.sip-router.org</a><br>
</div><a href=3D"http://lists.sip-router.org/****cgi-bin/mailman/listinfo/s=
r-****users" target=3D"_blank">http://lists.sip-router.org/**<u></u>**cgi-b=
in/mailman/listinfo/sr-<u></u>****users</a><<a href=3D"http://lists.sip-=
router.org/**cgi-bin/mailman/listinfo/sr-**users" target=3D"_blank">http://=
lists.sip-<u></u>router.org/**cgi-bin/mailman/<u></u>listinfo/sr-**users</a=
>><br>
<<a href=3D"http://lists.sip-router." target=3D"_blank">http://lists.sip=
-router.</a>**<u></u>org/cgi-bin/mailman/listinfo/*<u></u>*sr-users<<a h=
ref=3D"http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" targe=
t=3D"_blank">http://lists.sip-<u></u>router.org/cgi-bin/mailman/<u></u>list=
info/sr-users</a>><br>
><br>
<br>
<br>
</blockquote>
<br>
</blockquote><div class=3D"im">
<br>
<br>
______________________________<u></u>**_________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href=3D"mailto:sr-users at lists.sip-router.org" target=3D"_blank">sr-users=
@lists.sip-router.org</a><br>
<a href=3D"http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**user=
s" target=3D"_blank">http://lists.sip-router.org/**<u></u>cgi-bin/mailman/l=
istinfo/sr-**<u></u>users</a><<a href=3D"http://lists.sip-router.org/cgi=
-bin/mailman/listinfo/sr-users" target=3D"_blank">http://lists.sip-router.<=
u></u>org/cgi-bin/mailman/listinfo/<u></u>sr-users</a>><br>
<br>
</div></blockquote>
<br>
</blockquote><div class=3D"HOEnZb"><div class=3D"h5">
<br>
<br>
<br>
______________________________<u></u>_________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href=3D"mailto:sr-users at lists.sip-router.org" target=3D"_blank">sr-users=
@lists.sip-router.org</a><br>
<a href=3D"http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" t=
arget=3D"_blank">http://lists.sip-router.org/<u></u>cgi-bin/mailman/listinf=
o/sr-<u></u>users</a><br>
</div></div></blockquote></div><br></div>
--14dae9340e2beaa60e04b85070db--
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