When authentication times out that is one thing, but when it just fails like below Asterisk by default will not re-register until you the admin reload the sip or asterisk server:
voipserver*CLI> sip show registry
Host dnsmgr Username Refresh State&........
It is much more useful to have meaningful and detailed logging from tftp to see what is or isn't happening especially for VOIPand other embedded device appications:
Edit the file:
Change the server line like this:
server_args = -s /var/lib/tftpboot........
Cisco's CUCM (Cisco Unified Communication Manager) is a system that combines voice, video, data and mobile products into a single unified management suite. At its core, the CUCMis like a "Super PBX" that controls the flow of all communications through an organization even single or multiple site deployments.
Cisco's CUCMmakes communication more effective and simple through centralized management and unification of communications resources.........
I modified the default to the following for faster local dialing for North American area codes:
This is what I added to the above: "[2-9]xxxxxxxxxS0" so any 9 digit number is dialed instantly xxx-xxx-xxxx (the S0 at the end makes it dial right away). This makes dialing much quicker and is recommended.
The problem is that the default of most of these units is set for non-North American phones so the ring sounds like it cuts off and is not sequential.
How To Fix the Issue
Click on "Regional" and make sure you are in Advanced mode.
This problem seemed to happen recently but was likely causing issues before where the phone(s) do not ring.
Now there are a few reasons why this can happen especially if your adapter has DND mode enabled (disable it).
However that wasn't my issue and Ionly figured it out the other day when by fluke if you're on the phone (making a call) then calls will come in.
That's when Ifigured out the solution:
This likely app........
Sipura / Linksys PAP/VOIP/SIP Adapter Issue
Can't receiving incoming phone calls and you're behind a NAT router (99% of people)?
1. Login to the adapter.
2. Click on "Advanced" (location varies but usually somewhere on top)
2. Click on "SIP"
3. Scroll down to "NAT" (usually at the bottom).
You'll find 2 columns with 4 rows of drop-down boxes (they'........
NOTICE: rtp.c:1808 ast_rtp_read: Unknown RTP codec 100 received from
I've found that trying several times may work, but I also read changing your Sipura VOIP adapter settings as follows helps (but it still fails for me sometimes):
In adapter change the following under SIP/Advanced
Codec to G711a from G711u
Passthru Method from: NSE to ReINVITE
FAX CED Detect Enable: Yes
FAX CNG Detect........
Telus + 2Wire 2700 Router HorribleWell first of all let me say this is the only router/switch that sometimes seems to crash/disconnect computers on the local network.
This device also thought it would be smart to block VOIP packets coming from my Sipura ATA VOIP adapters so I disabled the [quote:cb89ba7bff]"Invalid TCP Flag Attacks (NULL/XMAS/Other)"[/quote:cb89ba7bff] option
Then all of a sudden I couldn't get onto any web pages, the wireless........
Asterisk Queue Context ExplainedThis was never explained in voip-info or any other site I read.
It is understood you can escape to a context from a queue and how to specify it.
What is NOT mentioned is that the context= you specify within the queues.conf refers to a [i:882f1e0aee]context that exists in extensions.conf[/i:882f1e0aee][/b:882f1e0aee]
This will save you headaches if you need to escape from the queue :)........