[Spce-user] routing calls between peers (Jon Bonilla (Manwe))
Ahmed Murad
am at sipalto.com
Mon Mar 25 07:45:23 EDT 2013
Hi Jon,
Many thanks for this info. For peering option 1 appears to be the one we
need. Billing is not an issue in this scenario and this spce is being used
as an SBC. We have 4 peers.
Peer 1 (inbound only), Peer 2(inbound only), Peer 3 & Peer 4.
Peer 1 sends inbound calls to the spce, we want to relay all Peer 1 calls to
Peer 4 which will then handle the call.
Peer 2 also sends inbound calls to the spce, we want to relay all calls
again to Peer 4.
Peer 3 and Peer 4 are our outbound Peers. Which subscriber calls will be
routed (inbound & outbound).
I can see the option ' force_outbound_calls_to_peer', how do I go about
setting the peer I want to force these calls to go to?
Again many thanks for your quick response, highly appreciated.
Best regards,
Ahmed
Ahmed Murad | SiPalto Ltd.
Tel: 0207 148 7525 | Fax: 0207 148 7526 | Email: am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website: www.sipalto.com
-----Original Message-----
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Today's Topics:
1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
4. Re: routing calls between peers (Jon Bonilla (Manwe))
5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
6. 483 Too Many Hops (Marc Rys)
7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))
----------------------------------------------------------------------
Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"
El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:
> Hello All,
>
>
>
> I tested sipwise yesterday.. install was successful very easy.
>
>
>
> Does anybody have problems with sip registration ?
>
> Why does it take 35sec to register.
>
>
>
> Also when it's registered and I dial takes long time to process call.
>
>
Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.
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Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"
El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:
> Hi.,
>
>
>
> I have "successfully" installed sipwise and am able to register users
> and make calls using any "normal" providers.
>
> Now
>
> In the same box, I have created a tunnel to another network that has a
> gateway.
>
> I am able to ping that gateway from the station that has sipwise also
> able to send traffic to the gateway, but I will get no answer because
> of the ip used.
>
> Let me put down the layout:
>
> All the following ips are fake but will clearly show the layout.
>
>
>
> Sipwise external ip: 200.200.200.200
>
> Sipwise internal ip: 10.0.0.1
>
>
>
> Tunnel router: 10.0.0.10
>
> Gateway across tunel: 192.168.10.10
>
>
>
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to
> gateway, the voice will come back thru tunnel because the gateway is
> replying to 10.0.0.1
>
>
>
> In my sipwise case, it does not work as sipwise is sending the call
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet
> and not the tunnel.
>
>
>
> How can I configure sipwise to send info to that gateway using ip in
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1> and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
>
>
Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?
You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine.
Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.
>
> The media relay is supposed to do that function but I have no clue on
> how to configure it.
>
> I could only create a socket with ip 10.0.0.1 and that allows me to
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
>
cheers,
Jon
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Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"
El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:
> Hello,
>
>
>
> I have been following your guides and successfully created & connected
> to a couple of external peers.
>
>
>
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
>
>
>
> But I get a timeout 500 error when I try to connect the softphone to
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
>
>
>
> Please could you point me in the right direction in regards to
> troubleshooting, is there anything I've missed?
>
>
>
Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.
Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.
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Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"
El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:
> Hello,
>
>
>
> I am new to SIPWISE, do you have any guides on how to route calls from
> 1 peer to another peer via the SIPWISE box?
>
>
>
> Any guidance would be greatly appreciated.
>
>
>
> Best regards,
>
>
>
Hi Ahmed
By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.
But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:
- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
preferences in the admin interface. If this option is set, any incoming
call
coming from this peer will be routed to pstn (to another or the same peer)
based on the peer rules. This will be done even if the callee is a local
subscriber.
- There's another option in config.yml which is called "allow_peer_relay".
This is global and will work like this: When a call comes from a peer it
will terminate the call locally if the calee is local or will send the
call
to the pstn following the peer rules if not.
The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration".
The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.
In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:
- Create your pstn gateways as peers. But other providers, you clients, the
ones you want to charge create them as *subscribers*. Creating them as
subscribers will allow you to rate them, apply ncos, prepaid (in case of
pro) and lots of features not present in peer options. You just need to
create a permanent contact for them, set them as trusted form the ip
addresses they use and allow all the clis (if you want).
Please let me know which scenario you're looking for and if any of these
three options match it.
cheers,
Jon
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Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"
El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:
> Good Day
>
> Is there any possibility that one could generate an Invoice to
> customers with a Trunk to SPCE 2.7?
You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.
>
> Is there any possibility to Generate a Bill to a Peering Partner for
> calls sent to SPCE 2.7. i.e Reverse billing?
>
Not at the moment. Peers can't be charged in this version of the spce.
cheers,
Jon
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Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
Content-Type: text/plain; charset="utf-8"
I've been evaluating SPCE over the last week. I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term. All of those test have worked very successful.
We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.
Any help is appreciated.
Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
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Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"
El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:
> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>
Please don't answer a mail to create a new thread. Use the "new mail" button
instead.
Cheers,
Jon
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