I'm so stumped on this. I've been scouring the forums for probably a month and a half now and tried changing ever configuration setting I can think of. Here's the story.

I've had good success running my Cisco 7940s and 7960s on Elastix 2.2.0.
All I typically have to is create extensions, use endpoint configurator to configure the phones, and then change the nat setting on each extension to no. If I don't change the nat to no, there will be x's next to the line on the Cisco phone which means it's not connected and not able to make or receive calls. That's my story with 2.2.0

I've been trying to upgrade to 2.3.0 with no success.
I've installed 2.3.0 from scratch. I've created 3 extensions. The 1st extension is for a cisco phone with nat = no, the 2nd extension is for a softphone (works perfectly), and the 3rd extension is for a different line on the cisco phone with nat = yes.
While in 2.2.0 as soon as I changed the setting to nat = no, the x went away and the phone worked, this is not the case on 2.3.0.

By the way, I can place and receive internal and external calls to softphones.

I would greatly appreciate help on this because it's really frustrating not to be able to update my server. Diagnostic information below:

yum update has been done.

sip show peers:
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
2301 (Unspecified) D N A 0 UNKNOWN
2302/2302 192.168.25.121 D N A 27168 OK (18 ms)
2303 (Unspecified) D N A 0 UNKNOWN

Note: I can force the Host's IP in which case the D under dyn disappears, the port becomes 5060, and the Status becomes OK, but that doesn't change the x next to the line on the cisco phone or its ability to place calls.

sip show peer 2301:
localhost*CLI> sip show peer 2301
* Name : 2301
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 2301@device
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "device" <2301>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 192.168.25.50
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,gsm:20,alaw:20)
Auto-Framing : No
Status : UNKNOWN
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No


Software package versions:

Kernel
Linux(i386)-2.6.18-308.8.1.el5

Elastix
package-elastix-is
elastix-asterisk-sounds-1.2.3-1
elastix-fax-2.3.0-3
elastix-addons-2.3.0-4
elastix-pbx-2.3.0-8
elastix-reports-2.3.0-4
elastix-vtigercrm-5.2.1-5
elastix-firstboot-2.3.0-7
elastix-framework-2.3.0-10
elastix-system-2.3.0-9
elastix-extras-2.2.0-1
elastix-im-2.2.0-2
elastix-a2billing-1.9.4-4
elastix-security-2.3.0-5
elastix-agenda-2.3.0-6
elastix-my_extension-2.2.0-6
elastix-email_admin-2.3.0-6

RounCubeMail
RoundCubeMail-0.3.1-12

Mail
postfix-2.3.3-2.3.el5_6
cyrus-imapd-2.3.7-12.el5_7.2

IM
openfire-3.7.1-1

FreePBX
freePBX-2.8.1-13

Asterisk
asterisk-1.8.12.0-0
asterisk-perl-0.10-2
asterisk-addons-1.8.12.0-0

FAX
hylafax-4.3.10-2rhel5
iaxmodem-1.2.0-1.1

DRIVERS
dahdi-2.4.1.2-11.1
rhino-0.99.4-3.rc1
wanpipe-util-3.5.25-0

I will make a suggestion here

Go to Freepbx gui and under system tools select asterisk sip settings, edit what you need and save

I have done that, sorry that I forgot to mention it.

I've tried both nat = yes, no, never, and route. I have the IP configuration set to dynamic, my host name entered, and my local networks specified.

Thank you for the suggestion. Any other ideas?

Did you find a solution to this?

Hi.

Did you compared the config file (SIPDefault.cnf) between 2.2.0 and 2.3.0?

Regards

Hi Did you manage to fix this I got the same problem with my cisco 7960 phone. Which was working under elastix 2.0.3

I think I did manage to fix my problem, but my results have been really inconsistent.
I'm pretty sure that an update to Elastix about 3 weeks ago helped with my most immediate problem (the 79XXs not registering with the PBX and therefore displaying x's next to each extension.

I have managed to get the x's to go away and the phones to register with my PBX successfully on fresh installs after about 3 weeks ago (not sure what was changed in the Elastix code, but obviously they fixed some sort of bug). I changed nat=no for each extension and that seemed to do the trick. I should note that I also fiddled around with the SIP settings in the Unembedded Free PBX (changed NAT=no there, then changed it back), but I think that updating to the lastest Elastix version via "yum update" and then changing nat=no on each extension should do the trick.

I'm actually now having intermittent "All circuits busy now" issues when dialing out, but not in, and I know that problem is completely unrelated, but still quite perplexing.

Good luck, I hope that helps. Post back if you're still having issues.

Hi Thanx for the quick reply.changing the nat setting from "never" to "no" allow me to make call from ciso phone to another open SIP hardphone but from the SIP phone to cisco didn't work. Cisco phone work both way when I use the 3cx softphone.
Before the upgrade I had a IP link with Avaya and NEC system but after the upgrade that link also not working it's through the all cicuit busy error when you try to make calls from asterix.
Thank you