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Hermenator

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Hi Guys,

I'm a newby to Operator and Asterisk. We have two offices with our PBX and SIP link at one office (Main) with a Forigate 60C firewall and another office (Remote) with extensions behind Smoothwall. These extensions need to register to the Main office. The PBX has two NIC's, one for internal network in range 192.168.26.0 and with fixed IP 192.168.26.150 / 255.255.255.0 and gateway 192.168.26.100 (firewall), the other with IP 10.20.1.4/255.255.255.248 and gateway 10.20.1.1. These are supplied by our SIP provider and registers on 10.8.0.5.

Extension 10 Main office (192.168.26.10) <--> PBX (192.168.26.150) <--> Firewall (192.168.26.100) <--> Internet with public address 41.161.39.184 <--> Extension 20 Remote Office on open network.

Used the help in setting up SIP behind NAT

This works fine for a while. For no apparent reason and no specific time we loose sound. As if the route for the calls get mixed up. When watching the call status in UI the call is registered but no sound and handshaking doesn't take place. Get the 3000s timeout error.

We thought this will be an easy setup (not like Trixbox, Elastix, Asterisk etc). What are we doing wrong?
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Jakub Zeman (Kerio)

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Hi,

did you follow this https://kb.kerio.com/article/configuring-nat-821.html knowledge base article?

Jakub
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ICT and Me

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@Herman,

Where are you from, Herman? Kerio Operator can real workm but it sounds there is a bandwidth problem.
I need some more information about your connection lines, how many extension in main office and remote office.

ICT and Me
Carlo Turk
The Netherlands
www.ictandme.nl
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Hermenator

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Hallo Jakub,

I used the setting as per the knowledge based article. I also tried setting the firewall with "open policies". Only with a virtual IP routing everything from outside through to the IP of the Kerio Connect box internally. The strange thing is that we had the system running perfectly yesterday afternoon for about 4 hours. The moment one of the clients, and I'm not sure which one, makes or receives a call it seems like the routing goes wonky. I then need to disable the internal interface, only keeping the SIP trunk interface active for us to be able to receive any calls, diverting them all to cellular numbers. Obviously then we cannot make any calls. While in this wonky mode I can see the actual call coming into Connect but no proper handshaking is taking place and the call will disconnect with a no answer error.

Hope this helps
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Hermenator

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@ICT and Me

I'm from South Africa. Our internet link is a permanent Microwave link running at 2Mb at the main office and a 1Mb ADSL link on the other side. We have 10 SIP channels coming into the main office with 23 extensions and 16 at the remote office.
The thing is I had this working perfectly yesterday afternoon running for about 4 hours. Without changing any settings except for SIP devices registering, SNOM 320's, XLite and SIPDroid, the system went wonky where incoming calls have no audio although the call is recognized in Connect but the IVR message not audible.
I have just tested it again. I disable internal NIC (192.168.26.150) (eth1). Eth0 still active and used to register with SIP provider (Neotel). They supply a PBX IP, Gateway and SBC IP for registering the SIP Trunk. Calls made to the office then works fine except obviously with no extensions. I divert all incoming calls to various cell phones, works fine. I then enable eth1 again. For a while the system then work correct after which the audio disappears again when making calls to the office.
???
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Vladimir Toncar (Kerio)

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Hi,

With the G.711 codec, the actual bandwidth consumption is about 80 kbit/s for one direction. So you are consuming about 0.8 Mbit/s in one direction for 10 concurrent calls. If there is other traffic, the line could get congested.

Vladimir
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alacratstore

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This is the great article to understand all the simple basic things about Kerio Operator like NAT configuration and it's working. Diagrams are very easy to understand and have clear out everything.

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