In this case, Iam assuming you have verified the firewall on either side, and the network conditions are not an issue (eg. no packet loss, no crazy jitter and normal latency). In some phones such as Grandstream or on your PBX, make sure you do NOT use the "adaptive jitter buffer" or similar settings. If possible disable that or change it to "fixed". If you do, you may wonder why when you first start a call the user cannot hear you or the voice s........

#Remember that you need a valid gateway IPunless the Asterisk server is on the same subnet and LAN Set Valid Gateway IP (if you don't have one already) ip route 0.0.0.0 0.0.0.0 GATEWAYIP Enable VOIPTrust voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 sip Set Credentials For Asterisk and Register To Asterisk sip-ua........

This is how we configure outside PSTNaccess or dialing through another SIPtrunk. Beware that this a simplistic example that neglects most security including SRTP First you'll need to be in config mode: Step - 1 Enter Voice Service VOIP security options There are more options but for now we'll just focus on security/allowing connections to and from our phones and the trunk. Router(config)#voice service voip ........

Have you checked your router/firewall logs and disconcertingly see connections to an unknown IP207.246.119.209:3478 from your Grandstream VOIPphones? You're not alone and the Grandstream forums have discussed this issue. However, even their own staff d........

Enable "cli" mode equivalent in JunOS cli Configure Mode configure So rather than going to the console on a Cisco switch and typing "enable" and then "conf t", the equivalent in JunOS is "cli" and "configure". How Do You Apply Changes You've Made? You can make all kinds of changes to the switch, but remember they are not........

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: vi /etc/xinetd.d/tftp 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: (*xx|[3469]11|0|00 [2-9]xxxxxxS0|[2-9]xxxxxxxxxS0|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) 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. *No........

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. ........

Dial **** Then 110# It will then say what your current IP address is.........

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[11389]: 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. [b:882f1e0aee] 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 :)........