2 msgMissing TRANSFER event in queue log when usingL...
2 msgsounds
3 msgRe: Determining the used codec for the IP Trunk...
5 msgCall Queues
2 msgsometimes half audio on 7960
4 msgCall Screening Not Working
1 msgIAX-Voicemail
2 msgIAX additional call-data
1 msgVisually impaired employees
2 msgSimple CDRs w/Asterisk/OpenSER.
6 msgREGEX expression for NXXNXXXXXX?
1 msgRe: exits in NJ
1 msgconnecting 1.2 and 1.4 using SIP

SIP / STUN / Network - Help!!
\ Gary (5 Jul 2007)
. \ Noah Miller (5 Jul 2007)

4 msgG729 on Solaris SPARC/x86/x64 Codec
1 msgProcess not draining UDP Recv-Q on port 5060
4 msgsometimes calls drop during attended transfer
3 msgAsterisk E1 card support Q.SIG
1 msgRe: AgentCallBackLogin vsAddQueueMember
1 msgRe: AgentCallBackLogin vs AddQueueMember
Subject:Re: SIP / STUN / Network - Help!!
Group:Asterisk-users
From:Noah Miller
Date:5 Jul 2007


 
Hi Gary -

> What I want to do is take one of my SIP devices to my office (which is ALSO
> behind another NAT) and try to connect with my home Asterisk box with it.
>
> For port forwarding, my AsteriskNOW box has a static IP on the inside of my
> NAT and I've configured the LinkSys router to port-forward ports 5060 (TCP &
> UDP) and all the RTP port range used (UDP only) to the static IP of the
> AsteriskNOW box. - Was this the right thing to do?

Yes. Just one thing, for the Sipura that you took to your office, did
you set "nat=yes" in sip.conf?

If the ports are forwarded, and Asterisk recognizes that this device
is NAT'ed, you shouldn't need to have a STUN server.


- Noah

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