3.0.0.4.334 firmware killed my SIP telephony and no ALG setting.

Discussion in 'ASUS N Routers & Adapters' started by j_modig, Jan 28, 2013.

  1. j_modig

    j_modig New Around Here

    Joined:
    Jan 28, 2013
    Messages:
    5
    Location:
    Uppsala, Sweden
    Hi all.

    First time poster and I hope this has not been covered before (i tried searching).

    I recently upgraded my N56U to the latest available ASUS firmware ( 3.0.0.4.334). Unfortunately it killed my SIP-based telephony (provided by ISP).

    I have set up port forwarding for UDP port 5060 to my ATA-box on the LAN side. I get my connected DECT phone to ring if I call it but I can't pick up the call.

    I had this exact problem on an older D-Link router and sorted it out by disabling SIP ALG (application layer gateway). I can, however, not seem to find a setting in the current ASUS firmware.

    Any input on this is very much appreciated.
    TIA
    /Johan
     
  2. Log in / Register to remove this ad

  3. Nerre

    Nerre Senior Member

    Joined:
    Oct 17, 2012
    Messages:
    313
    Normally you would need to either have your ATA use upnp or forward the RTP ports (the ports used for audio, your ATA probably can be configured to what ports it should use).
     
  4. j_modig

    j_modig New Around Here

    Joined:
    Jan 28, 2013
    Messages:
    5
    Location:
    Uppsala, Sweden
    Thanks for your input Nerre.

    I'll double check those settings but I have made no changes in the ATA since last time it was working properly with my original firmware (not sure what exact version but it was 1.0.xxxxx something).

    Right now it does not even establish a SIP-session (and I do not get dial tone trying to call out).

    Cheers
    /Johan
     
  5. Nerre

    Nerre Senior Member

    Joined:
    Oct 17, 2012
    Messages:
    313
    For our Siemens S675IP I have set up the router to give it a static IP (so the forwarding is set up for that IP).

    Then I forward:

    5060 tcp+udp
    5004-5024 upd

    to that IP.

    Those ports are set up in our Siemens S675IP setup.

    (I have an RT-AC66U, but used the same configuration on my old router that was an old Pentium machine running M0n0wall.)
     
  6. j_modig

    j_modig New Around Here

    Joined:
    Jan 28, 2013
    Messages:
    5
    Location:
    Uppsala, Sweden
    Thanks for the detailed input Nerre.

    I'll keep tinkering when I get home from work.

    My ATA is a Ping Communications NPA201 (http://www.pingcom.net/products/voicecatcher/) and my ISP specifies no port forwarding settings other than 5060 UDP.

    And I also use the static IP option in my N56U to make sure the MAC of my ATA keeps the same IP address on my LAN.

    Cheers
    /Johan
     
  7. RMerlin

    RMerlin Part of the Furniture

    Joined:
    Apr 14, 2012
    Messages:
    14,044
    Location:
    Canada
    An ATA shouldn't need any forwarded port since it's the one establishing an outbound connection. I have no forwarded port here for my Cisco SP112 ATA.

    You can disable the SIP helper by running these commands over telnet:

    Code:
    nvram set nf_sip=0
    nvram commit
    
    Then reboot the router to apply the change.

    Not sure they will help in your case (I keep it enabled here).
     
  8. j_modig

    j_modig New Around Here

    Joined:
    Jan 28, 2013
    Messages:
    5
    Location:
    Uppsala, Sweden
    Thanks for the tip RMerlin. I'll give it a try.

    Maybe I'm using the term ATA incorrectly but how would my "SIP-box" (for lack of better words) detect incoming calls without port forwarding?

    There is of course the option of putting my ATA in front of the N56U since it can act as a router by itself but I really don't like that idea.

    Obviously I need all the help I can get here ;)

    Cheers
    /Johan
     
  9. j_modig

    j_modig New Around Here

    Joined:
    Jan 28, 2013
    Messages:
    5
    Location:
    Uppsala, Sweden
    Problem solved!

    This worked perfectly! I now have both up to date firmware and fully functional telephony.

    Many thanks RMerlin.

    Cheers from Sweden
    /Johan
     
  10. RMerlin

    RMerlin Part of the Furniture

    Joined:
    Apr 14, 2012
    Messages:
    14,044
    Location:
    Canada
    In an actual ATA's case, your ATA registers with your ISP's SIP server (typically an Asterisk server), and will refresh its registration on a regular basis (mine re-registers every 2 mins if I remember correctly), keeping an outbound connection to port 5060 constantly open.

    It's possible that box your ISP provides isn't a pure ATA but actually a SIP server, in which case it might explain the need for a forwarded port AND having to disable the SIP helper (the latter is required if you run your own Asterisk server, for instance).

    In any case, what matters is that you got it working :) Would probably be a good idea for Asus to add an option to the official firmware to enable/disable the SIP helper. As you can see, the code is already in the firmware, it's only missing a webui option to toggle it.
     
  11. Nerre

    Nerre Senior Member

    Joined:
    Oct 17, 2012
    Messages:
    313
    But when the call is placed the audio is transmitted using UDP on other ports than 5060, and if those ports are not forwarded the incoming audio won't reach the ATA.

    I guess the SIP helper tries to do that forwarding automatically?
     
  12. BCSteve

    BCSteve Occasional Visitor

    Joined:
    Mar 23, 2013
    Messages:
    43
    RMerlin... could I ask how you knew about the telnet command? Is there a list somewhere for what settings, etc, are available via telnet on this router?
     
  13. RMerlin

    RMerlin Part of the Furniture

    Joined:
    Apr 14, 2012
    Messages:
    14,044
    Location:
    Canada
    I know about it because I have been developing a custom firmware for the RT-N66U and RT-AC66U for the past year, so I'm fairly familiar with the source code. Toggling the state of the SIP helper is something I have added to my webui a few months ago, specifically for the issue described here.
     
  14. alvino

    alvino New Around Here

    Joined:
    Apr 1, 2013
    Messages:
    1
    Thanks rmerlin for the solution. Works for my a580ip Siemens phone. Almost wanna give up on this newly purchased router.
     
  15. woland

    woland Occasional Visitor

    Joined:
    Feb 2, 2013
    Messages:
    14
    Running 3.0.0.4.270.26 firmware.
    When tried to use SIP in android devices the audio did not came from outside.
    After issuing set nf_sip=0 everything started to work perfect.
    Thanks for solution.
     
  16. JohnnyBeGood

    JohnnyBeGood Occasional Visitor

    Joined:
    Mar 18, 2014
    Messages:
    40
    Location:
    Seattle, WA
    I have Elastix v2.4.0 and I had no audio issue when connected remotely via android Zoiper app. After disabling this I can hear again :) Thanks!
     
  17. kelso

    kelso New Around Here

    Joined:
    Sep 9, 2014
    Messages:
    2
    Hey Merlin, quick question (sorry to bump an old thread). Is the sip helper you toggle from CLI the same as the SIP Passthrough that's in the Web GUI (Advanced / WAN / Nat Passthrough)?

    I've been deploying hosted pbx service for a long time, and this is the first time using a dark knight. When I disable sip passthrough, I cant get the phones to register to our SBC, but with it enabled, they register. I just know that in general ALG causes us huge headaches and I wasnt sure if those were the same things or not.

    Thanks in advance.
     
  18. RMerlin

    RMerlin Part of the Furniture

    Joined:
    Apr 14, 2012
    Messages:
    14,044
    Location:
    Canada
    The webui option controls whether the conntrack helper module for sip gets loaded by the kernel or not. Same thing as the old nf_sip setting, which no longer does anything.
     
  19. kelso

    kelso New Around Here

    Joined:
    Sep 9, 2014
    Messages:
    2
    Any ideas on why disabling the SIP Passthrough would kill the ability for my phones to pass registration info through to our SBC?
     
  20. RMerlin

    RMerlin Part of the Furniture

    Joined:
    Apr 14, 2012
    Messages:
    14,044
    Location:
    Canada
    That option might be incorrectly named since it's not a passthrough option, it's a conntrack helper function. Enabling the option will tell the firmware to load the conntrack helper module, which is designed to help SIP clients communicate through NAT.

    So in your case, enabling this option is actually required, since you do have SIP clients that need the helper.
     

Share This Page