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Title: SIP remote phones Post by: mdraghici on Friday 19 June 2009, 04:28:28 am Hello
we have been using EFW for several years now and we are extremely happy with this solution. Congratulations to the team for a great product! last week we updated to EFW 2.2 and we have been unable since to configure our remote phones to work properly. We are using a Trixbox PRO server behind the firewall and we forwarded all required ports as per the official documentation and our previous setup, namely: Code: TCP ANY : 8000 => 192.168.0.23 : 8000 Trixbox VPN the outgoing firewall is disabled, and SIP proxy is enabled in transparent mode. The same setup has been working properly before with EFW 2.1 the diagnostic tool from the Trixbox PRO shows that connection can be established on all ports except for port 5060. However, remote phones manage to register to the Asterisk server. Whenever we initiate a call to/from a remote extension - the phones ring but there is no audio either way. I am completely baffled by this and am running out of options. Does anyone have any ideas on what we can try? I am confused on whether we need to use source NAT (set to NO NAT) and whether SIP proxy should be on. Since the problem seems to occur on 5060 I am assuming this is somehow related to the SIP proxy intercepting communication. However, the phones ring each other which I undertand it as a connection being established. Any ideas would be highly appreciated. Thanks in advance for your help. best regards, ~mircea Title: Re: SIP remote phones Post by: ego on Saturday 20 June 2009, 07:14:53 am I had a similar problem when I replace Ipcop with Endian 2.2
I have a little phone system for use of our family. We have Elastix 1.3.2 running virtual in my office, two sip3102, one on my home in Miami and other in my home in Spain. Elastix is working behind Ipcop and all work fine. I need replace Ipcop with Endian. Yesterday, I did it, but the phones didn't get hear the sounds that asterik was playing, and I had to return to Ipcop. I am investigating what was wrong, I opened all outgoing traffic from green network to red w/out success, also activated sip proxy. The registration was susscesfull and I could see the route for the calls in asterisk, the problem was with rtp protocol. Title: Re: SIP remote phones Post by: mithun on Wednesday 20 January 2010, 07:28:31 am Very similar is my problem. one way audio on all outgoing calls. the called telephone cannot hear my voice. I have configured Dnat and Snat, still no luck. Any clues?
Title: Re: SIP remote phones Post by: Saltee on Monday 22 February 2010, 01:22:05 am have you tried using IAX, it encapsulates RTP and only utilises one port. In real life your phones may not support IAX though.
what does the Asterisk console say when calls are established - type asterisk -r ++++ at the console of your Asterisk (Elastix in your case) server, this will bring up the Asterisk CLI is verbose mode and you'll see what's going on. also - make sure the extensions are setup as being behind a NAT router - this is often forgotten. good luck. Title: Re: SIP remote phones Post by: Steve on Tuesday 23 February 2010, 03:01:02 pm ................. UDP ANY : 5060 => 192.168.0.23 : 5060 Trixbox SIP ................. the outgoing firewall is disabled, and SIP proxy is enabled in transparent mode. The same setup has been working properly before with EFW 2.1 the diagnostic tool from the Trixbox PRO shows that connection can be established on all ports except for port 5060. However, remote phones manage to register to the Asterisk server. ..... Wouldn't you want to disable transparent mode since you are forwarding port 5060? Just a thought. :) Title: Re: SIP remote phones Post by: jamrod on Tuesday 13 April 2010, 07:15:57 pm I am also having this problem with EFW2.2
It effects calls that I make from remote SIP extensions and also the confererences lose audio when one or more calls from the SIP trunk are connected. I did not get this with IP Cop and a DLink router. |