Re: [asterisk-users] sip and extensions

2012-07-06 Thread Shitian Long
Hello, If you would like to make out bound call (from Asterisk to SIP provider), it is fine. But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this sip.conf register = 5552530146:your_password@sip3.voipvoip.com/5552530146

Re: [asterisk-users] sip and extensions

2012-07-05 Thread Tim Nelson
- Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146

Re: [asterisk-users] sip and extensions

2012-07-05 Thread Thomas Perron
Hi, I changed these codes to not coincide with actual account info. Thanks On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register =

Re: [Asterisk-Users] sip phone extensions at a remote site

2005-04-13 Thread Cameron Beattie
If you're using SIP I think what you want is canreinvite=yes which means the two remote user clients can talk directly to each other. Asterisk disappears from the loop which means no accounting. I think NAT causes problems in this scenario also. More details on the wiki Regards Cameron -