Extension Refuses to Register - Once it does, another refuses - Rinse, Repeat

PBX Firmware: 1.812.210.57-1
PBX Service Pack: 1.0.0.0
Your Linux Distribution: (Redhat CentOS release 5.7 (Final))
Your FreePBX version: (2.10.0)

I have a problem with extensions failing to register with the server. I’m certain the phones are set up properly as I will fumble around with them, reset the device over and over and reset the router - without changing settings - and they will eventually come up (and the next extension stops working).

It feels like there is a connected device limit but in all my reading and searching through FreePBX and astrisk documentation, I can’t find a mention of it or any problem like mine. I’m hoping that’s because it’s a novice thing and me, being the novice I am, am missing it.

Here’s a screenshot of my System Status screen (you’ll notice the 1 phone that isn’t registered):
http://dl.dropbox.com/u/1923731/attachments/Screen%20Shot%202012-06-13%20at%207.10.59%20PM.png

Here is a snippet of code with some messages that I can’t seem to find - not sure if they relate or not:

[2012-06-13 17:03:27] VERBOSE[2912] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/fpbx-2-deb42490-0000088e [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7450-0000088f connected line has changed. Saving it until answer for SIP/fpbx-2-deb42490-0000088e [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7452-00000890 connected line has changed. Saving it until answer for SIP/fpbx-2-deb42490-0000088e [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7457-00000891 connected line has changed. Saving it until answer for SIP/fpbx-2-deb42490-0000088e [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7450-0000088f is ringing [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7452-00000890 is ringing [2012-06-13 17:03:27] VERBOSE[2912] app_dial.c: -- SIP/7457-00000891 is ringing [2012-06-13 17:03:47] VERBOSE[2952] chan_sip.c: -- Got SIP response 302 "Moved Temporarily" back from 192.168.1.3:5060 [2012-06-13 17:03:47] VERBOSE[2912] app_dial.c: -- Now forwarding SIP/fpbx-2-deb42490-0000088e to 'Local/[email protected]' (thanks to SIP/7450-0000088f) [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Not accepting call completion offers from call-forward recipient Local/[email protected];1 [2012-06-13 17:03:47] NOTICE[2912] chan_local.c: No such extension/context [email protected] while calling Local channel [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Forwarding failed to dial 'Local/[email protected]' [2012-06-13 17:03:47] VERBOSE[2952] chan_sip.c: -- Got SIP response 302 "Moved Temporarily" back from 192.168.1.52:5060 [2012-06-13 17:03:47] VERBOSE[2912] app_dial.c: -- Now forwarding SIP/fpbx-2-deb42490-0000088e to 'Local/[email protected]' (thanks to SIP/7452-00000890) [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Not accepting call completion offers from call-forward recipient Local/[email protected];1 [2012-06-13 17:03:47] NOTICE[2912] chan_local.c: No such extension/context [email protected] while calling Local channel [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Forwarding failed to dial 'Local/[email protected]' [2012-06-13 17:03:47] VERBOSE[2952] chan_sip.c: -- Got SIP response 302 "Moved Temporarily" back from 192.168.1.58:5060 [2012-06-13 17:03:47] VERBOSE[2912] app_dial.c: -- Now forwarding SIP/fpbx-2-deb42490-0000088e to 'Local/[email protected]' (thanks to SIP/7457-00000891) [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Not accepting call completion offers from call-forward recipient Local/[email protected];1 [2012-06-13 17:03:47] NOTICE[2912] chan_local.c: No such extension/context [email protected] while calling Local channel [2012-06-13 17:03:47] NOTICE[2912] app_dial.c: Forwarding failed to dial 'Local/[email protected]' [2012-06-13 17:03:47] VERBOSE[2912] app_dial.c: == Everyone is busy/congested at this time (3:0/0/3) [2012-06-13 17:03:47] VERBOSE[2912] res_musiconhold.c: -- Stopped music on hold on SIP/fpbx-2-deb42490-0000088e

You need to change the voicemail uri to *97 (or whatever you changed it to).

Most routers NAT mangles SIP. If the router has any SIP helper or ALG functions turn them off.

Most firewall/routers today have VPN capability I would use it. It takes all the problems out of remote SIP. You can also hand out the tftp server address and remote provision the phones.

I would not use any type of bridging, why waste bandwidth on broadcast traffic over a WAN?

I have a SIP client on my iPhone/iPad. I set it up for the very extension that isn’t working currently and it’s working remotely. It would seem there may actually be a network issue… but I have no idea what that would be. I’ll keep researching it. Any other thoughts/suggestions/ideas are welcome in the meantime.

** edit ** After posting that I saw your response… thanks, I’ll look into it.

Are the phones remote? Sounds like a NAT issue.

Also call forwarding is set on the phone as you are getting diversion messages.

Lastly, somehow you have the term voicemail in the dial field with the diversion header and it is trying to dial that SIP URI.

SkyKing - Hey, you helped answer my questions about my ip2007 phones previously… thanks for helping again!

Currently, it is affecting a remote phone (my company owns the building next door as well and I have the phones there nat-ed into our system in the adjacent building). It started on a local phone though. Assuming they are unrelated, do you have any thoughts as to what is going on? I have verified that all four of my remote phones’ settings are the same (with exception of line specific settings). Nat is allowed on all 4 extensions in freepbx. I’m actually really familiar with networking so if this is something outside of a setting in the phone system, I can’t see what it is. Maybe I’d be better off trying to bridge the connections and make it all internal rather than remote? Seems like it should work regardless

My IP2007 phones use “voicemail” as the default code to enter voicemail… that may be what is causing that issue. I will also check on the call forwarding.

I appreciate all those answer… wasn’t trying to clutter up on thread with three things.

Conclusion… although a strange one imo…

It did seem to be a network problem although that problem may have been in the firmware of my router. I updated my linksys wrt56gl router to a dd-wrt.com firmware in anticipation of creating a vpn connection and suddenly the problem cleared up. I do plan on completing the vpn connection but it’s fixed in the meantime.

Skyking - You’re my hero. Thanks so much for helping!

Just a thought…had you run out of addresses in your DHCP pool?

BF

w5waf - Thanks for the idea!

Not according to the settings… if the router was hung up, that’s a different story. I had 200 available ip’s and am using just over 100 of them. Upon updating the firmware, my settings were all carried over (with the exception of my port forwarding - which I had to re-setup - easy enough). That was the first thing I looked at when I realized the problem as network troubleshooting is easy enoughand freepbx is a new project for me… a new learning curve.

Are you trying to register the same extension with multiple different phones?

alan_mousty: Nope. Each phone had it’s own extension.

As a Side note to that question: I was able to register the same extension on my iPhone and iPad (sip client) along with my line at work - all at the same time… it didn’t cause a problem but I turned it off as it was unnecessary and I didn’t want to experience a problem later on. Just did it to test it out and see if I could.