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Asterisk: Sip phone Linksys wip330

i have a linksys wip330 phone. Im tring to get it to work outside my network. I watch the asterisk console as i try to connect to it via the remote network service and i see the phone information register in the asterisk console but the phone itself says its not ready. I attempt to make a call any way and i see the console show asterisk going thru the montions and the pstn or extension will even ring but there is no sound in either direction. what can i do to fix this?

here are some configs

sip.conf

canreinvite=no
localnet = 192.168.1.0/255.255.255.0
Nat = yes
externhost = topdat.serveftp.net
bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68


rtp.conf

[general]
rtpstart=10001
rtpend=20000

linksys router with upd and tcp port 5060 pointed to internal ip of server and udp port 10001 - 20000 point to the same.
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jonnydollar
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jonnydollar
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2 Solutions
 
grbladesCommented:
Do you have a STUN (proxy) server configured on the phone?

Have a look at http://www.voip-info.org/wiki-STUN. There are some public STUN servers listed there aswell.
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jonnydollarAuthor Commented:
no i dont

what do i do to get that working?
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grbladesCommented:
Generally you just configure one of the public server onto the phone. The STUN proxy detects if the phone is behind any sort of firewall or a device which performs network address translation and attempts to fix any problems this may cause which typically are one way audio or no audio at all.
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jonnydollarAuthor Commented:
i find that the registation of the phone seems not to be fully setup. heres why. i mentioned that i have a linksys wip330 phone that has been unsuccessful with remote connects thus far. But to test would use my exten software with a sip account setup and point to the remote voip server as well. I watch the asterisck cli and i see the notice that the sip account registerd, but on sxten itself it says that the login failed: request timed out. however there is enough information passed for setup and teardown of a call without the audio.

is this something the stun server might resolve or could smething additional be happening?
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feptiasCommented:
That sounds like an example of the SIP request for registration reaching the server, but the response (200 OK) not being able to get back to the phone that sent the request. What IP address is asterisk sending the 200 Ok back to?

By the way, is the phone behind its own NAT device? And Asterisk is behind another? If so, have you set port forwarding on both NAT devices or only one?
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jonnydollarAuthor Commented:
the ip address of the phone is gonna be a private one and also behind a nat device (linksys or netgear etc router). I am only about to config properties on the nat device i own. I wont be able to control wat another router is configured with. would this be required for the phone to work?
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grbladesCommented:
Being behind NAT should not affect the ability to register. You are sending a SIP message so a reply should automatically be permitted back through the NAT device. Normally the problem you will encounter with the phone being behind NAT is that there will be one way audio (when using the phone you cant hear them but they can hear you) and this can often be fixed by using a STUN server.
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feptiasCommented:
The phone might be able to work if you can enable STUN on it, but the situation you described is one of the more difficult ones - i.e. the phone is behind NAT, the server is behind a different NAT and you cannot set port forwarding on one of the NAT devices.

I've written some notes on my web site about SIP and NAT that might be of interest:
http://www.feptias.co.uk/AdviceSIPbehindNAT.htm
http://www.feptias.co.uk/AdviceSIPNATSolutions.htm
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jonnydollarAuthor Commented:
ok this wip330 phone does not have a STUN server field. would it be somethin else like outbond proxy or register proxy?
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grbladesCommented:
Yes its also commonly known as a proxy.
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jonnydollarAuthor Commented:
ok the phone has these fields
proxy ip
register proxy ip
outbound proxy ip

which of those should the stun server be placed
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feptiasCommented:
The "outbound proxy" is the sip server that your phone sends requests to when it is making a call. The "register proxy" is the sip server that your phone registers with so its location is known and it can receive inbound calls. Both of these would need to be the Asterisk server in the scenario that you have described. Sometimes, one IP address can be hosting both STUN server and sip proxy server services. Unfortunately, Asterisk cannot act as a STUN server (as far as I know).

The address of a STUN server can be provided in an SRV record in DNS, but there is no guarantee that your wip330 would read that information and use it even of the SRV record was present. The DNS domain that would be used is the same one that you specify in the phone for your user registration. e.g. User_ID=456701@mysipdomain.com.  For more info about SRV records in DNS view this link:
http://www.voip-info.org/wiki-DNS+SRV

There is a reference to STUN support being added to the wip330 in this link:
http://www.voip-info.org/wiki/view/WIP330
 
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jonnydollarAuthor Commented:
excellent help. after i updated the phone for STUN support. I pointed it to a STUN server and have been able to register.
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kanebearCommented:
This issue can be as much about the firmware version on the WIP-330 as anything else. I do not have a STUN server but successfully register to and use Asterisk from all over the world with my WIP330 from NATted IPs to an Asterisk PBX that's behind a NATted firewall.

In at least one firmware revision, the default SIP port was set to 6060. This required configuration through the web interface (HTTP://IP_address_of_the_phone, default user :admin default pass:0000) as not all settings were available from the phone interface (still the case). Once that was rectified, everything worked fine.
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