site Search:


 
    All Forums Hot Topics Gallery






how-to block ads


 
Search Topic:
Uniqs:
290
Share Topic
Posting?
Post a:
Post a:
Links: ·ALL ·Review Your VoIP Provider ·VoIP Providers ·VoIP FAQ ·Porting Rules ·What Codec?
AuthorAll Replies

fuzzuy

join:2006-11-02
North York, ON

[Equipment] RTP300 internal call trasnfer??

Hello,

I have been a silent reader of this forum for a long time now and learned a lot tricks from here so first like to thank every one for that and now I have questions may sound stupid to few here but I will ask.

Here what I am trying to achieve I have my RTP300 configured for freephoneline.ca service on line 2 which is working fine now I have callcentric account where I do I have a US DID which I like to configure on line 1 and get that transferred to line2 internally is it possible ?? as I know I can use PBXES and sipsorcery etc but that thing is just not working for me and I believe freephoneline.ca block all incoming sip calls, so my best bet is to registered both of them in RTP300 and get them transferred any help would be appreciated thanks in advanced.

Stewart

join:2005-07-13
kudos:14

FPL won't accept a SIP call, but you can send it directly to the ATA. In your router, forward the UDP port to the RTP300, which corresponds to the setting for SIP Port in your RTP300 Line 2. I believe that the default is 5061. Then, if you don't have a static IP address, you'll need a dynamic DNS name (set up in your router or in a PC dynamic DNS client). For initial testing, you can use your current public IP address, so you won't have to mess with dyndns.

In your Callcentric preferences, go to DID forwarding and set the DID to forward to e.g. 19051234567@123.45.67.89 (the part before the @ must match the value of User ID for Line 2; the part after is your public IP address or dynamic DNS name). From another phone, call your Callcentric number to test.

Alternatives to the above:

1. Get a two-line phone, or replace the RTP300 with an IP phone with multiple lines.

2. Forward the Callcentric DID to your FPL number (will cost $0.0198/min.)

3. Get a free US DID from VoxOx and forward it to your FPL number (free, but you won't be able to choose the US number).

4. Post details about your trouble with PBXes; perhaps we can help.


fuzzuy

join:2006-11-02
North York, ON

Hi Stewart,

Thanks very much for the quick reply here i took your advice easy one off course i got the new number from voxox and forwarded to my FPL number and that works thanks again.

Now about pbxes issues is i got it configured and see the both online and configured the extension also and routed to the one which is registered to my ATA but when i call both numbers they ring but no ring on ATA but still i can see the log in the PBXES log there but in reality at ATA there is no ring so i gave up that hope as i was suspecting that FPL does not allow sip calls so i gave up.


Stewart

join:2005-07-13
kudos:14

I assume that your PBXes setup has one extension and two trunks, one registered to FPL and one to CC.

Under Inbound Routing, you should have only the default "/" route, which should specify the extension.

Calls to either the CC or the FPL DID should ring the ATA. Does calling the CC DID work as expected? What does PBXes Call Monitor show for a failing call?


fuzzuy

join:2006-11-02
North York, ON

I has two extensions and two trunks and i will try what you said here and feed back here thanks


Monday, 04-Jun 02:50:47 Terms of Use & Privacy | feedback | contact | Hosting by nac.net - DSL,Hosting & Co-lo
over 12.5 years online © 1999-2012 dslreports.com.
Most commented news this week
Hot Topics