Search results

No personal experience with TW, but unless your ISP is oversold in your neighbourhood, you won't be able to tell the difference between VoIP over internet and your PRI. If you're sharing the internet link with other internet traffic, make sure your router is configured to throttle both incoming...

If your device is mission critical, you may want to invest in a loop current regulator to protect it: http://www.sandman.com/loop.html#LoopCurrentRegulator I have phone line -> surge protector -> loop current regulator -> OBi110.

Here's a 26-minute conversation between Lenny and an astonishingly unkind telemarketer:

No ducks? :(

Unfortunately no. You're correct that the rate centre would not change. If a service provider can't port it now, porting to a cell won't change anything. The exception to this rule is Google Voice, which only ports from cell phones.

Someone was having audio issues this morning. The call arrived, but I did not receive any RTP. I tested via SipBroker and direct SIP URI and both worked for me, so I'm not sure what the problem was. As long as it is working properly for you now, that is fine.

It doesn't work for me either. You're welcome to use [email protected] until the other one is working again, if you want to! You must pass your Caller ID number. If you use the SIP URI, you must also use g.711.

You may want to check rtp.conf for rtpstart and rtpend, by default 10000-20000. You can use your router to forward this UDP port range to your PBX. Many routers will not require this, but perhaps yours does. 10000 ports is a great deal, far more than you need unless you are handling thousands...

I am still able to check the DID using the following code: same => n,GotoIf($[${REGEX("1xxxxxxxxxx" ${SIP_HEADER(TO)})}=1]?context,extension,priority)

I bet that's why it works. Can anyone explain the difference between "anonymous SIP" and "SIP guests", to confirm? Edit: confirmed. Found this: I checked Callcentric's suggested configuration for PIAF. They suggest that you add context=from-pstn to sip_general_custom.conf. My...

The outbound call from the OBi to Asterisk is via SIP. Not via the LINE port. Per the manual, this option allows you to "select a method to detect if an outbound call on the PSTN line has been answered". Since the OP is not making an outbound call on the PSTN line, this setting is unlikely to...

The OP is having problems with inbound calls.

Hi PBXEHR, how is the troubleshooting going?

I can't see any problems there. In that case, I would recommend you continue with troubleshooting and inspect the console in verbose mode when a call arrives. This will prove or disprove my theory, and give you an idea of what is going on when the problem is happening. Just to check - you're...

I also use the OBi110 for this purpose. I have never observed an OBi answer a call, without a phone going off hook, and without the AA being invoked. Is it possible your PBX is the one answering the call? My LINE Port InboundCallRoute is simply {sp1(incoming)} . incoming is a valid extension...

If you don't mind plain vanilla Asterisk code, you can get Lenny here: http://www.dslreports.com/forum/r29474609-

I can help with that if necessary. (I'm sure MacNix's friend is more than capable, but in case he's too busy or something.)

I would LOVE to hear the results of that. :)

My PBX is behind a Tomato router (iptables firewall) that operates "restricted cone NAT". In other words, the router will only allow incoming traffic from a service provider I've recently registered to. I do have anonymous calls enabled, though that's not a security risk as unwanted SIP traffic...

No offense taken! I was hoping I could learn a little about how other service providers do it as the ones I use all use DNS for load balancing and/or failover. It is currently a moo point as Callcentric is down.