懒人发问: 一个Freephoneline 的问题

也可能直接用italkbb的人买。
 
  1. Remove and disable the "Outbound Proxy" settings. Set "Use Outbound Proxy" to NO.
    Now, the primary proxy is voip.freephoneline.ca (or, voip2.freephoneline.ca), and No Outbound Proxy (empty entry).
  2. Now, you needed to specify a STUN server to use. I'm using: stun.counterpath.net as mine.
    To set a STUN server, navigate to the SIP tab on your SPA112, SPA122, SPA2102, PAP2T, or SPA504G phone. At the bottom of the SIP configuration page, you will see a spot to define the STUN server. Enter stun.counterpath.net or stun.3cx.com, or some other STUN server you may prefer. Then set the STUN Enable to YES.
Performing steps 1 and 2 above has solved my "No Inbound Audio" problems with the new freephoneline.ca servers.
 
  1. Remove and disable the "Outbound Proxy" settings. Set "Use Outbound Proxy" to NO.
    Now, the primary proxy is voip.freephoneline.ca (or, voip2.freephoneline.ca), and No Outbound Proxy (empty entry).
  2. Now, you needed to specify a STUN server to use. I'm using: stun.counterpath.net as mine.
    To set a STUN server, navigate to the SIP tab on your SPA112, SPA122, SPA2102, PAP2T, or SPA504G phone. At the bottom of the SIP configuration page, you will see a spot to define the STUN server. Enter stun.counterpath.net or stun.3cx.com, or some other STUN server you may prefer. Then set the STUN Enable to YES.
Performing steps 1 and 2 above has solved my "No Inbound Audio" problems with the new freephoneline.ca servers.
这个 MARK 一下,下次出问题再参考。 现在已经工作了就不折腾了:)
 
这个 MARK 一下,下次出问题再参考。 现在已经工作了就不折腾了:)

我的也有问题了,别人打不进来,一直占线。
 
我的也有问题了,别人打不进来,一直占线。

VoIP Unlock Key Credentials

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Support
posted this on Feb 07 12:53

Version 1.01
The Freephoneline VoIP Unlock Key provides sip credentials that can be used to configure any SIP client to work with the Freephoneline service. Please review the following Freephoneline guidelines to set up your SIP client. Failure to follow the required guidelines will result in account suspension followed by a notification email. Once your configuration adheres to the guidelines, service will be restored.
These guidelines may change over time - if and when they are changed freephoneline users will be notified at the email address used for account login a minimum of 7 days prior to the changes being required. Please ensure your contact information is accurate by visiting your account profile on freephoneline.ca.
Required Settings

SIP Server:
voip.freephoneline.ca
Alternative SIP Server:
voip2.freephoneline.ca
Transport:
UDP
Port:
5060
Notes:

  • It is always best to use the DNS name for your SIP server as our infrastructure is always expanding/changing/being maintained. The IP addresses which you register to will change over time.
  • Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended.
Registration Interval:
3600 seconds (1 hour)
Registration Expiry:
3600 seconds (1 hour)
Failed Registration Re-Try Interval:
120 seconds
Recommended Settings

STUN/ICE:
Disable
NAT Mapping Enabled:
Yes
NAT Traversal:
Enable sending Keep-Alives only:
  • on Grandstream HT-701 ATAs this setting is “no, but send keep-alive”
Keep Alive Message:
NOTIFY or a UDP PING Packet
  • For Linksys/Cisco devices, use ‘Nat Keep Alive Msg’ = $NOTIFY or $PING
  • Never use REGISTER as your Keep Alive message
Keep Alive Interval:
20 seconds*
*Audio may be affected if this value is adjusted
Notes:

  • The above settings can be used to configure your SIP client to function in common home network configurations. Since there a thousands of home network configurations, it is impossible for us to provide a single set of parameters that will always work. As a VoIP Key purchaser, it’s expected that you have knowledge of your network and how to configure your SIP client properly.
  • Freephoneline does not offer STUN server. However, you may use a public one if your wish.
RTP Settings

Supported Codecs:
g711-u (uLAW) and g729
Suggested RTP Packet size (psize):
0.020 - This ensures audio packets every 20 milliseconds, achieving better quality (trade-off: bandwidth)
Notes:

  • The above settings are used by your ATA to determine how the audio will be encoded/decoded across the Freephoneline network.
 
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