SIP to PSTN Calls through VOIP Provider.

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Anonymous

I want to forward incomming sip calls to PSTN through voip provider which I already configured for extension 100, but when I am trying to contact ext 100, getting status below and unable to connect.


Server Status:

13:29:43.406 Incoming c5 "101"<sip:[email protected]:5061> <sip:[email protected]> Incoming call (before routing)
13:29:43.437 Destination Ext:100 has been found for Ext:101, but it hasn't been registered! Unable to complete call.
13:29:43.437 Rejected c5 "101"<sip:[email protected]:5061> <sip:[email protected]> Call destination cannot be resolved
13:29:43.437 Terminated c5 "101"<sip:[email protected]:5061> <sip:[email protected]> Call ended


Also as I know, no need to conifure any softphone or voip phone for ext 100, because i am using ext 100 with voip provider, so when other extension will dial ext 100 their call will connect to PSTN No. through configured VOIP Provider. Am I right ?

If No, please give me some details how my sip calls will forward to PSTN through VOIP Provider, I dont want to use any physical devide or phone.
 
Hi,

I am sorry if I misunderstood your question but it seems you simply want to place a call to the PSTN through a VoIP Provider. You will need to;

1. Create and extention
2. Configure a sip phone to use that extension
3. Purchase a an account (sign up) from a VoIP Provider.
4. Configure the 3CX PhoneSystem to make use of the 'VoIP Provider'

On creation of the 'VoIP Provider' the PhoneSystem will create a default dialing rule, this will allow you to make extenal calls using a 0 prefix. Simply dial 04499888888 to call 4499888888 on the PSTN.

Hope you manage to get what you want working, let us know if you run into any difficulties.
 
Extension

Hi,

Extensions and VOIP providers are separate things. You can not configure an extension for a VOIP provider. You configure an extension, then configure a VOIP provider and then configure an outbound rule which will divert the call via the VOIP provider....
 
Call routing doesn't work with me

Hi
I have installed a xten Voip UA at extention 100. I made a rule which diverts all call starting with 0 to voip buster. I created a VoipBuster link which registers. When I dail an existing number which starts with a "0" the call log doenst show any interaction with the VoipBuster Registrar It does show the correct phobne numbers and local IP addesses but not invites going to the VoIPBuster Registrar
Have I missed somthing or is the routing not working. At leased I expected interaction with the VoIPBuster registrar. Can any one give me a help

18:20:15.901 Terminated c23 "Oscar Goos"<sip:[email protected]> "0365401235"<sip:[email protected]> Call ended
18:20:15.901 Rejected c23 "Oscar Goos"<sip:[email protected]> "0365401235"<sip:[email protected]> Call destination cannot be resolved
18:20:15.901 CallConf::findDestination: Found destination Dummy for caller Ext:100
18:20:15.901 Registrar::resolveAddress: Registrar can not resolve "0365401235"<sip:[email protected]>
18:20:15.885 Registrar::checkAor: Registrar resolved sip:[email protected] as <sip:[email protected]:61792;rinstance=e6707339e615abc9>
18:20:15.885 Incoming c23 "Oscar Goos"<sip:[email protected]> "0365401235"<sip:[email protected]> Incoming call from 100
 
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