Got a Cisco Sipura SPA 941 hardware phone and I am connected to Asterix (most likely) VoIP server of my provider.

I am behind a firewall (ADSL) and I have put my phone to the DMZ zone, so it should be reachable from the internet.

Now I can make calls without any problems, but when somebody calls me I can hear it ringing, but when I pickup the earphone, nothing happens and I see "Answering Call" on the display. On the other side of the line I still hear its ringing.

Now question to all superuser VoIP experts: what option should I search on my phone to chage to get this working? My friend gave me advice DMTF (got Auto), tried all options without any luck.


Even in a DMZ, the firewall can affect sip signaling, or there may be issues within another device down the line. For example, you may have the phone in a DMZ on your own router, but your ISP has a routed modem and that device is selectively blocking some of the sip signaling (ALG, SPI, Intrusion Detection) which is not allowing the phone to acknowledge to the server that it has picked up the call, even though locally you have. This is very very very common. I have even seen instances where the ISP has an ALG setup on their core routers that has intermittently blocked the ACK signal from the phone on an invite, causing that exact problem.

To put it simply, make sure that ALL devices have ALG, SPI, and Intrusion detection disabled. Also make sure that cisco specific ports are open in all devices between your phone and the voice server. This may require a call into your ISP to clear the entire path (if you have your own router and your ISP is able to bridge the modem, that is usually a huge help).

If your voip provider can get you the call ladders for the issue, I can take a look and tell you where the signaling is failing (though they really should be able to tell you that too).

  • Many thanks for tips. I know what is intrusion detection, not sure about ALG, SPI. Will try to find that on my router. My ISP tells me to open the following ports: TCP a UDP 5060, RTP 10000-20000 – lzap Aug 15 '11 at 11:17
  • So I found ALG section there, I only have "SIP" enabled there. So according to your advice I am disabling it right now. There's also intrusion thing ("Enable DOS and Portscan Protection") but I have it turned off. – lzap Aug 15 '11 at 11:22

If your phone rings and you pick up the receiver, you can see on the Asterisk CLI that it thinks the phone has been answered, and yet you hear nothing on the phone. The RTP part of the call is being blocked somewhere, most likely the firewall in your case.

If you have VoIP phone service from your ISP, you may find that all RTP ports have been stopped by your ISP's device for its own use. In that case you will need to define a different RTP port range on Asterisk for your SIP Truck service.

  • It should be noted that this a 4 year old question. – Burgi Jun 21 '16 at 8:21

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