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[SOLVED] Disabling SIP on Drayteks for Asterisk etc.

  • ghenry
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23 Sep 2009 17:30 #7 by ghenry
Replied by ghenry on topic Re: [SOLVED] Disabling SIP on Drayteks for Asterisk etc.



Yes to all, apart from changing all "Sip Accounts" ports to 5081
Every works fine IF the ATA's (the built in ones) are not active i.e. set to port 0 as soon as I set them both back to 5060 (for my voip accounts) I can no longer log into my Asterisk box from the outside - via my IP address. The ATA's have nothing to do with the Asterisk side of things at the moment.

One thing I have just noticed is if turn off the ATA's (set to port 0) for a moment to let a soft-phone login to the Asterisk box THEN set the ATA's back to 5060 everything works ok. Its a pain but seems to work, not very satisfactory though ! If I then log out then try to log back in it times out error 408 - from eyebeam soft-phone



What you can do here then is change either the ATA ports or Asterisk to listen on others in order to run both. As soon as 5060 is enabled on the router you can't forward it obviously. Choose different and adjust the outside calls appropriately.

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24 Sep 2009 11:07 #8 by myozone
Replied by myozone on topic Re: [SOLVED] Disabling SIP on Drayteks for Asterisk etc.

ghenry wrote:



Yes to all, apart from changing all "Sip Accounts" ports to 5081
Every works fine IF the ATA's (the built in ones) are not active i.e. set to port 0 as soon as I set them both back to 5060 (for my voip accounts) I can no longer log into my Asterisk box from the outside - via my IP address. The ATA's have nothing to do with the Asterisk side of things at the moment.

One thing I have just noticed is if turn off the ATA's (set to port 0) for a moment to let a soft-phone login to the Asterisk box THEN set the ATA's back to 5060 everything works ok. Its a pain but seems to work, not very satisfactory though ! If I then log out then try to log back in it times out error 408 - from eyebeam soft-phone



What you can do here then is change either the ATA ports or Asterisk to listen on others in order to run both. As soon as 5060 is enabled on the router you can't forward it obviously. Choose different and adjust the outside calls appropriately.



That would work however, If I change the Asterisk ports I have to remember to do this every time I change soft-phone's etc. I can't change the ATA's as these connect to my DID voip providers, I could add them to Asterisk trunks but that's not really what I want to do. The only other way is to use external ATA's ! - just to fix a issue with the router.

Thanks again

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05 Oct 2009 15:25 #9 by macdaddy
Replied by macdaddy on topic [SOLVED] Disabling SIP on Drayteks for Asterisk etc.

I have just noticed is if turn off the ATA's (set to port 0) for a moment



How did you do this? Did you go into the SIP Accounts section and change the port to 0 there?

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05 Oct 2009 15:29 #10 by myozone
Replied by myozone on topic [SOLVED] Disabling SIP on Drayteks for Asterisk etc.

macdaddy wrote:

I have just noticed is if turn off the ATA's (set to port 0) for a moment



How did you do this? Did you go into the SIP Accounts section and change the port to 0 there?



Yes, In VOIP setup > SIP Related Functions Setup > altered 'SIP Port' to zero 0 in both ATA's 1 and 2 this is on a 2600VG BTW

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05 Oct 2009 15:55 #11 by macdaddy
Replied by macdaddy on topic [SOLVED] Disabling SIP on Drayteks for Asterisk etc.
Thanks for your reply. We too had a Vigor 2600 that worked OK with our Asterisk behind it, but that router doesn't have ADSL 2+. We are now with a 2820vn and can't get any incoming or internal calls to work at all. We've tried ghenry's suggestions as well as paying for premium support and it's still a no-go :(

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21 Jan 2011 22:04 #12 by dannyboy1121
Replied by dannyboy1121 on topic Re: [SOLVED] Disabling SIP on Drayteks for Asterisk etc.

ghenry wrote: Hi All,

Just paid for premium support, but here's the answer from them for the benefit of others.

Telnet to your router and do:

sys sip_alg 0

Also, go to VoIP and even though all your "SIP Accounts" are empty, go through each (12 on a 2820vn) and set the SIP port to say 5081 and click ok. Do this for all and above. Also change your RTP ports to finish at port 9999.

Then point your "Open Ports" on the NAT setting to your Asterisk server:

SIP/UDP 5002-5080
RTP/UDP 10000-20000

There you go!

Gavin.

http://www.suretectelecom.com



This worked for me with my Trixbox configuration. I tried for ages to get it up and running but could never see calls getting forwarded to Asterisk/Trixbox regardless of port forwarding rules. SIP_ALG didn't fix it alone - but as I run with a 2710Vn, I followed the advice to alter all the ports for each of the SIP accounts away from 5060 to 50xx.

Straight off the bat, it's started working - forwarding calls to the Trixbox. So even if I didn't have anything configured on them - I still needed to change them off port 5060.
Dan

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