Setup FreeSwitch behind Kamailio Dispatcher.
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
Hi, If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything. On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi,
If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb , else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too.
Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK].
Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work.
In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy.
I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch.
Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later.
My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch.
But when going through Kamailio Dispatcher it fails between the extensions.
So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it.
My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi,
If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb , else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too.
Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK].
Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work.
In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads. On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote: Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy.
I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch.
Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later.
My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch.
But when going through Kamailio Dispatcher it fails between the extensions.
So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it.
My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi,
If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb , else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too.
Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK].
Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work.
In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string" value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough.
I would like to have as little load on Kamailio as possible, as it just should load balance.
Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again,
Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache)
Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort.
By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup!
-- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> Hi Sammy.
I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch.
Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later.
My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch.
But when going through Kamailio Dispatcher it fails between the extensions.
So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it.
My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote: Hi,
If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too.
Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK].
Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work.
In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Any chance you can post your opensips config? I've been struggling with mixed success trying same setup. On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string" value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote:
Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com ;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string"
value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc.
And
calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb ,
else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with
FreeSwitch
behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to
work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello Vik and Matt, Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work. Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy. kind regards, Leon On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote: Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string" value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc. Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer? I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS ! On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij <leon@scarlet-internet.nl>wrote:
Hello Vik and Matt,
Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work.
Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy.
kind regards,
Leon
On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote:
Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com ;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string"
value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all
it
does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb ,
else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with
FreeSwitch
behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> > > I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. > I've setup Kamailio via > this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html > I've installed FreeSwitch from scratch on Ubuntu > via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start > Now, when registering extensions via Kamailio Dispatcher I'm able to > call to FreeSwitch and listen to hold music. But that's it. I'm not able to > call between extensions etc. > Can anyone help me setting up FreeSwitch to accept registration, calls > etc. from Kamailio and everything else that is needed to use FreeSwitch > behind a load balancer? > I'm very new to FreeSwitch, but I'm trying to use the terminal (without > any GUI etc.) as I want the installation to be as clean as possible. So I > would prefer very precise help, as I'm still getting hold of FreeSwitch.
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello Sammy, Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase. Kind regards, Leon On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote:
Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS !
On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij <leon@scarlet-internet.nl> wrote: Hello Vik and Matt,
Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work.
Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy.
kind regards,
Leon
On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote: Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string" value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up. 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> > > I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. > I've setup Kamailio via > this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html > I've installed FreeSwitch from scratch on Ubuntu > via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start > Now, when registering extensions via Kamailio Dispatcher I'm able to > call to FreeSwitch and listen to hold music. But that's it. I'm not able to > call between extensions etc. > Can anyone help me setting up FreeSwitch to accept registration, calls > etc. from Kamailio and everything else that is needed to use FreeSwitch > behind a load balancer? > I'm very new to FreeSwitch, but I'm trying to use the terminal (without > any GUI etc.) as I want the installation to be as clean as possible. So I > would prefer very precise help, as I'm still getting hold of FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
:) :) :) LOL-- right- Sorry, I'm from Narnia didn't know. Original Poster had issue with Kamailio and FS setup, even if both the projects share common parents but I don't believe that sharing opensips.cfg will help the issue here. Anyways I appreciate the spirit to help others. -- BR, Sammy On Tue, Nov 15, 2011 at 7:05 PM, Leon de Rooij <leon@scarlet-internet.nl>wrote:
Hello Sammy,
Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase.
Kind regards,
Leon
On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote:
Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS !
On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij <leon@scarlet-internet.nl>wrote:
Hello Vik and Matt,
Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work.
Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy.
kind regards,
Leon
On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote:
Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com ;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/ sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string"
value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all
it
does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this >
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb ,
else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> > > Hi everyone. In regards to my earlier question regarding with FreeSwitch > behind Kamailio Dispatcher, I've attached a call from extension 1001 to > 1002, which fails. It just hangs for some time and then says that 1002 > cannot be found, and then the voicemail for it comes up. > 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com
>> >> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >> I've setup Kamailio via >> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> I've installed FreeSwitch from scratch on Ubuntu >> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> Now, when registering extensions via Kamailio Dispatcher I'm able to >> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >> call between extensions etc. >> Can anyone help me setting up FreeSwitch to accept registration, calls >> etc. from Kamailio and everything else that is needed to use FreeSwitch >> behind a load balancer? >> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >> any GUI etc.) as I want the installation to be as clean as possible. So I >> would prefer very precise help, as I'm still getting hold of FreeSwitch. > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Just gave the config I'm working on because two ppl asked for it. But from what I understand from the original poster, is that he's having trouble with calling from one ua to another where both ua's are registered on fs and all signaling originating from the ua's towards fs is balanced by a proxy - which is the same as what I'm trying to achieve and have basically working now. I agree that just pasting a config is not the solution for the OP as there are so many combinations / possibilities to implement this. So indeed as you asked: * what is the topology * do you want the proxy to _always_ be in the signaling path between fs and ua The trace that was attached shows: U 2011/11/13 17:08:28.788249 216.231.104.54:5060 -> 76.33.21.131:5060 INVITE sip:1002@sip.my-domain.com SIP/2.0. Record-Route: <sip:216.231.104.54;lr=on>. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. From: <sip:1001@sip.my-domain.com>;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: <sip:1002@sip.my-domain.com>. U 2011/11/13 17:08:28.788575 76.33.21.131:5060 -> 216.231.104.54:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 216.231.104.54;branch=z9hG4bKcfcd.3c288a3.0. Via: SIP/2.0/UDP 96.224.14.164:41264;rport=41264;branch=z9hG4bKPjXqQE4ca7HrI0tOo8umTYyui8DbpSPjDQ. Record-Route: <sip:216.231.104.54;lr=on>. From: <sip:1001@sip.my-domain.com>;tag=.0D5vpFns0g9Byw0Q9cYBm0v9IqMjj6A. To: <sip:1002@sip.my-domain.com>. U 2011/11/13 17:08:28.803526 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002@96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. From: "Extension 1001" <sip:1001@76.33.21.131>;tag=FgF5QDyZjQXtr. To: <sip:1002@96.224.14.164:43975;ob>. Contact: <sip:mod_sofia@76.33.21.131:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. U 2011/11/13 17:08:29.805308 76.33.21.131:5060 -> 96.224.14.164:43975 INVITE sip:1002@96.224.14.164:43975;ob SIP/2.0. Via: SIP/2.0/UDP 76.33.21.131;rport;branch=z9hG4bKy6m7184yKFDXr. From: "Extension 1001" <sip:1001@76.33.21.131>;tag=FgF5QDyZjQXtr. To: <sip:1002@96.224.14.164:43975;ob>. Contact: <sip:mod_sofia@76.33.21.131:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d93ed90 2011-11-11 20-17-21 -0600. etc I can assume that: * 76.33.21.131 is fs * SIP packets (for 1002) are sent from FS directly to 96.224.14.164:43975 Since the invite towards the ua is such a high port, can I assume that the ua is nat'ed ? The INVITE is sent multiple times without an answer, so I think that the UA (1002) behind 96.224.14.164 has registered through the proxy on the fs server, so a nat entry (triple internal ip/port - local ip/port - external ip/port) in the state table of the firewall in front of the ua exists towards the proxy where it registered to, so the fs server can never directly reach the UA, so the INVITE from FS towards the UA must go through the proxy as well which isn't happening right now - that can be fixed by setting the outbound-proxy param on fs in the sip_profile. Anyway, am looking forward to updates on this as I really want to get this working as well ! Kind regards, Leon On Nov 15, 2011, at 4:06 PM, Sammy Govind wrote:
:) :) :) LOL-- right- Sorry, I'm from Narnia didn't know.
Original Poster had issue with Kamailio and FS setup, even if both the projects share common parents but I don't believe that sharing opensips.cfg will help the issue here. Anyways I appreciate the spirit to help others.
-- BR, Sammy
On Tue, Nov 15, 2011 at 7:05 PM, Leon de Rooij <leon@scarlet-internet.nl> wrote: Hello Sammy,
Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase.
Kind regards,
Leon
On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote:
Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS !
On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij <leon@scarlet-internet.nl> wrote: Hello Vik and Matt,
Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work.
Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy.
kind regards,
Leon
On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@gmail.com> wrote: Any chance you can post your opensips config? I've been struggling with mixed success trying same setup.
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij <leon@scarlet-internet.nl> wrote:
Hi, I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module. First I used a simple route script in opensips with using dispatcher, but after the first message (from ua through proxy to fs), the proxy would get out of the signaling path, while I want it to stay in. To fix that, I added record routing in the proxy configuration so it stays in the path. Registrations are also balanced towards the fs servers - but some clients are nat'ed so the contact header was wrong - fix_nated_contact() in route and in onreply_route fixed that. To get calls originating from fs to go through the proxy before going to the client, I tried: originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string: originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park which seemed to do the same thing (does anyone know if there's any difference between the two variants ?) Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example: <param name="dial-string" value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> so now it's possible to just call the user as: originate user/some_user@mydomain.com &park nice ! And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating: <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/> so now you can leave out the sip_route_uri or fs_path variables, it just works. My proxy config still needs a lot of work, but I got the basic functionality working, still need to find out about using acls on fs side based on a sip header added by the proxy - I'm already adding X-AUTH-IP headers on messages from clients towards fs which should do the trick, but I didn't test it yet. Anyway, would love to read some configs of ppl who successfuly setup opensips/kamailio/openser/... proxies in front of fs, what their experience is with using record-routing or not, etc. regards, Leon
On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
The handling of several FS are not a issue, actually easy enough. I would like to have as little load on Kamailio as possible, as it just should load balance. Also, having to handle users on both Kamailio and FS makes unecessary work loads.
On 14/11/2011, at 00.32, Sammy Govind <govoiper@gmail.com> wrote:
Hi again, Why don't you just let Kamailio handle registrations. Anyway I was thinking about that LBing the registration would result in such a scenario that calling one extension to another would not make a successful call because the other endpoint maybe registered on some other FS. This may further lead you to making a dial-plan which would work somewhat like DUNDI but it'd just have to search all the FS servers before joining a call.(Obv there are other intelligent approaches to minimize the headache) Try following the link I sent you and implement that in front of your FS, I think that Kamailio configuration is so well written that anyone can start understanding kamailio and implement such setups with little effort. By explaining your topology I meant how do you plan to use Kamailio in front of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a topo-hiding or SBC like setup! -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi Sammy. I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch. Currently I only have 1 FreeSwitch, for testing this basic setup. Next move would be 2 FreeSwitch with 1 common database etc. But that's later. My FreeSwitch is the basic setup, without anything else. Just as the installation manuel is written. So I have my extension 1000 - 1019 etc. And calls between them works when connected directly to FreeSwitch. But when going through Kamailio Dispatcher it fails between the extensions. So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it. My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.
On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hi, If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy
2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> > > Hi everyone. In regards to my earlier question regarding with FreeSwitch > behind Kamailio Dispatcher, I've attached a call from extension 1001 to > 1002, which fails. It just hangs for some time and then says that 1002 > cannot be found, and then the voicemail for it comes up. > 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com> >> >> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work. >> I've setup Kamailio via >> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html >> I've installed FreeSwitch from scratch on Ubuntu >> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start >> Now, when registering extensions via Kamailio Dispatcher I'm able to >> call to FreeSwitch and listen to hold music. But that's it. I'm not able to >> call between extensions etc. >> Can anyone help me setting up FreeSwitch to accept registration, calls >> etc. from Kamailio and everything else that is needed to use FreeSwitch >> behind a load balancer? >> I'm very new to FreeSwitch, but I'm trying to use the terminal (without >> any GUI etc.) as I want the installation to be as clean as possible. So I >> would prefer very precise help, as I'm still getting hold of FreeSwitch. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
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Henrik Aagaard Sørensen
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Leon de Rooij
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Matt Stockton
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Sammy Govind
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Vik Killa