Re: [Asterisk-Users] Asterisk-to-Asterisk call setup problem (one way works fine)
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote: Tor Houghton wrote: PHONES1=IAX/[EMAIL PROTECTED] Did you try IAX2/[EMAIL PROTECTED] ? Erm, no. Haha, I cannot believe I spent days trying to fix that. It works! My internal asterisk took the call! Yay! Thanks! Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-to-Asterisk call setup problem (one way works fine)
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote: Tor Houghton wrote: PHONES1=IAX/[EMAIL PROTECTED] Did you try IAX2/[EMAIL PROTECTED] ? Actually, I think I found the culprit. It seems (ho hum), that the IAX softphone re-registered (reinvited?) with the external IAX server, so that the translations in the NAT gateway got muddled. (Incidently, do I need to forward udp/5036 and udp/4569 on the NAT gateway when the internal Asterisk registers with the external Asterisk? Or would I only need this if it were the other way round?) Cheers, Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-to-Asterisk call setup problem (one way works fine)
Hi,
I'm having a bit of a problem. I have two Asterisk servers, one serving SIP
clients on the outside of a NAT, the other on the inside. The internal one
also serves PSTN and IAX clients.
When I call someone (who is on SIP) from any phone registered with the
internal Asterisk, I get through to them no problem. The issue is when they
try to call me; for some reason they do not get routed (I am assuming the
extensions are wrong).
The outside Asterisk logs this:
Mar 13 00:18:38 NOTICE[6052352]: app_dial.c:545 dial_exec: Unable to create channel of
type 'IAX'
== Everyone is busy at this time
The outside * extensions.conf contains
PHONES1=IAX/[EMAIL PROTECTED]
[sip]
exten = 2201,1,Ringing
exten = 2201,2,Dial(${PHONES1},20,Ttm)
exten = 2201,4,Hangup
and the outside's iax.conf is so:
[inside]
context=iax
type=friend
secret=PASSWORD
host=dynamic
tos=nodelay
qualify=yes
trunk=yes
Is there something I've missed here? (I'm suspecting I am, but I can't seem
to find any hints on how to fix it.)
Hope someone can help. Been banging my head against this for a few days
without much luck.
Thanks,
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-to-Asterisk call setup problem (one way works fine)
Tor Houghton wrote: PHONES1=IAX/[EMAIL PROTECTED] Did you try IAX2/[EMAIL PROTECTED] ? -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
