We have an internal SIP server using Asterisk in our office, all of our staff use these phones without issue (inside the office).

What we would like to do is have it so people with external SIP capable devices can also use your SIP server.

I have setup the SIP features in our Astaro, with the SIP server been our SIP servers public facing IP address.

I have then made the allowed SIP networks, ANY.

When I attempt to use a SIP device that is out on the internet I can see in the Asterisk logs that it registers but then straight away disconnects, its like it has only incoming communication with the SIP server.

Any suggestions on how this should be setup?

Thanks

From the online Manual in the Astaro "The Session Initiation Protocol (SIP) is a signalization protocol for the setup, modification and termination of sessions between two or several communication partners. It is primarily used in setting up and tearing down voice or video calls. SIP uses TCP on port 5060 to negotiate which dynamic port range is to be used between the endpoints when setting up a call. Since opening all ports within the dynamic range would cause a severe security issue, the firewall is able to handle SIP traffic on an intelligent basis. This is achieved by means of a special connection tracking helper monitoring the control channel to determine which dynamic ports are being used and then only allowing these ports to pass traffic when the control channel is busy. For that purpose you must specify both a SIP server and a client network definition in order to create appropriate packet filter rules enabling the communication via the SIP protocol."

What I find is that it is successfully completing the following: "the firewall is able to handle SIP traffic on an intelligent basis. This is achieved by means of a special connection tracking helper monitoring the control channel to determine which dynamic ports are being used and then only allowing these ports to pass traffic when the control channel is busy." As of now, I can get my external SIP device to call into my Astersik server, and and the internal SIP device rings, you pick up.... but no voice/audio. This says to me that the signaling from the external device to the asterisk server is fine but when the Asterisk server hands over the phone call to connect it between the two phones this is failing. Which is what I understand the Astaro features are meant to do.... have I got this right?

your help is appreciated.

Hi,
you need to add some packet filter rules to allow the various port ranges your VoIP system uses out. You can exclude SIP of course.

Ian M

__________________
Home Power User unlimited licence - v7.50x - AMD X2 5050e with 2gb,1 intel NIC, the onboard NIC and netgear gs108t with vlans.

that machine has full access outbound......

I have been playing around with Trixbox and Astaro this weekend. I initially setup VOIP security on the firewall. I was able to make calls, but they would drop after a short period of time. Always within 1 minute. After checking the Trixbox config I finally turned off VOIP security and just left my packet filter and NAT rules enabled and everything started working just fine...

Not exactly sure what VOIP security does, but in my case it was causing issues.

David

SIP support is going to be fixed in 7.402 which will be released very soon.

__________________
"Datenautobahn: Einrichtung zur schnellen Übertragung großer Datenmengen (z.B. über das Telefonnetz)" (DUDEN, 21. Auflage)

Mario Schmidt
QA Engineer
Astaro AG

Does someone have a list of trixbox rules that need to be added since the sip proxy doesn't appear to work?

You can create packetfilter rules for whatever ports are needed, and DNAT also, if needed.

Look at your packetfilter logs for clues as to what ports the Trixbox is trying to use, or look at the docs for Trixbox.

Barry

__________________
http://DealBert.net
Home & business end-user since v1.x

• ASL 6.3x, HP DL145 Dual Opteron, 1GB RAM, 6 gigE NICs, 50-IP Platinum License
  
• ASL 7.3x, Dell PE1550 Dual PIII 1GHz, 1GB RAM, 2 NICs, 50-IP Platinum License
  
• ASL 7.5x, 17-watt fanless mini-ITX system: MSI IM-945GSE-A Atom n270, 2GB RAM, Morex T3310 case. 2 Intel GigE, 3 VLANs. 80G 5200rpm 2.5" HD
  Netgear GS108T gigE VLAN switch & Linksys WRT54G WAP
  Total network infrastructure: 27 watts. 100-IP Home User. FiOS 10mb/2mb

I am having a similar issue w/ SIP not working and have tried version 7.402 and now 7.403 without success. Can anyone verify a working install with the IP PBX being internal and external users accessing it via SIP?

Thank you,
Dave