The sipXecs mailing list has been extremely helpful for me. I'll forward this 
along to the list and try to get some input from the experts there. I was going 
to post the configuration to the wiki if we were successful anyway. 

To those on the list reading this: This is an attempt to configure a SIP trunk 
from a AsteriskNOW box sitting behind a NAT to a sipXecs box also sitting 
behind a NAT. The sipXecs box is inside the DMZ. The AsteriskNOW box has 
5060-5080 TCP/UDP and 10001-11000 UDP forwarded to it. sipXecs responds with 
either 401, or 405 messages regardless of our configuration parameters so far. 
I contacted FreePBX's paid support for help.

About the 408 messages: Yesterday I noticed a communication problem between our 
primary and our secondary sipXecs servers. I removed the secondary from the 
server list and forced all extension registrations to the primary. Last night 
the registrations expired and I was getting the 408 timeouts from all 
extensions--even those registered from the external side. I noticed that inside 
/etc/sipxpbx/registrar-config the line SIP_REGISTRAR_SYNC_WITH : <secondary 
server here> was still there. I commented the line and then performed a 
"service sipxecs restart" which cleared up the problem.


-----Original Message-----
From: FreePBX Support [mailto:[email protected]]
Sent: Thu 6/11/2009 9:28 AM
To: Matt Keys
Subject: [SUPPORT #CJF-510965]: sip trunk from freepbx to sipxecs 4.0
 
Matt, 408 means we are still not quite connecting somewhere along the line. I 
am not sure why sipXecs in using the rport of 1229? It was using 5060, then 
1229 started showing up? Did something change on the sipX?

Were you taking about the extensions on the sipX or the couple that are on the 
.170?

After trying several iterations of trunk configuration, I am not sure we are 
going to figure this out without some sort of feedback from the sipXecs box.

As I am curious to know if .170 can make use of a different provider, I got an 
account from sipgate.com which has a free DID and 200 minutes out free right 
now. The sipgate trunk registered and can receive calls.

With no changes from me we are now seeing:

Reliably Transmitting (no NAT) to 208.71.234.166:5060:
REGISTER sip:pbx.pfnnet.net SIP/2.0
Via: SIP/2.0/UDP 208.71.234.170:5060;branch=z9hG4bK3ded8829;rport
From: <sip:[email protected]>;tag=as1c9ca41a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 146 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7062914527", realm="pfnnet.net", algorithm=MD5, 
uri="sip:pbx.pfnnet.net", nonce="1f4aa885992bbc38a3ee8571071759474a310165", 
response="1f20ab8735e4a2e1e8f54acd4be97fce"
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
mauerpbx*CLI> 
<--- SIP read from 208.71.234.166:5060 --->
SIP/2.0 401 Unauthorized
From: <sip:[email protected]>;tag=as1c9ca41a
To: <sip:[email protected]>;tag=755e4ed2
Call-Id: [email protected]
Cseq: 146 REGISTER
Via: SIP/2.0/UDP 208.71.234.170:5060;branch=z9hG4bK3ded8829;rport=1229
Www-Authenticate: Digest realm="pfnnet.net", 
nonce="944df37c118c5d8fd8fa77e7b2fe98074a310179"
User-Agent: sipXecs/4.0.1 sipXecs/registry (Linux)
Date: Thu, 11 Jun 2009 13:07:05 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu, path
Content-Length: 0

Last night I set sip_nat.conf to:

nat=no
externip=208.71.234.170
localnet=192.168.1.0/255.255.255.0

As I suspect that sipXecs is going to require that. We need to figure out a way 
look at the sipXecs a bit closer.

Lastly, I am not going to be available until tonight. If you need someone to 
look at this further today, say so and I'll see what I can do to get someone to 
step in.




---

Robert Keller
FreePBX Support
To Learn More About FreePBX Training Please Visit
http://freepbx.org/open-telephony-training-seminar

Ticket Details
===================
Ticket ID: CJF-510965
Department: Hourly Support
Priority: Normal
Status: Open


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