Thread: Link2VoIP
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Unread 11-08-2007, 02:32 AM   #6
emoci
 
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Quote:
Originally Posted by Badweather View Post
So I called in via the sipbroker pstn in my area. What I've done is unreged my softphone so the system is going to Voicemail right away. My DID is just blank sound, although I know it answers because I get a Voicemail notification. The sipbroker in works fine and I actually hear my voicemail box. So the problem appears to be between Voxalot and Link2VoIP. What woud you suggest I try in for this? Link2Voip's support is trying to help me but unfortunately we don't have access to logs with Voxalot to see how the connections are being made in order to debug.
1. Do exactly as you did, but leave the SoftPhone Connected and see how you hear yourself (or better yet get someone else on the phone while you talk from the SoftPhone)

2.Or go to SIPBroker - EziDial , then enter your VoXalot SIP URI 123456@voxalot.com (again replace 123456 with your VoX number). Now a call should come in from ECHO test

3.Things to do to Improve incoming issues:

-Preferably make sure you have STUN set up (stun.xten.org has been working for me)--however I don't know if the SoftPhone will have an option for this.
-If you are behind a router that has UPnP, make sure it's active
-This is optional, but it may help to open these port ranges and forward them to your ATA's IP(in your case to the IP of the computer running the SoftPhone):

5050-5064
5000-5005
16300-16500 (This is a bit hardware/software specific, it may change from one to the next, I would try doing the first two and see if they make a difference)

4. When trying to register the number initially the Optimize Audio Setting should be set to No (and the Symmetric NAT handling on the member page is set to Yes). If you do see improvement up to this point, then follow this guide to perform audio optimization:
http://forum.voxalot.com/voxalot-sup...e-quality.html

5. Finally, did you try setting up codecs as ulaw;alaw;g726;g729;ilbc;gsm


Quote:
Originally Posted by martin View Post
This thread should shed some light on your problem:

http://forum.voxalot.com/voxalot-sup...html#post10291

Unfortunately unless the embedded provider adheres to the SIP RFC there is not much we can do
I could be wrong, but if Link2Voip indeed had this problem wouldn't the SipBroker Access Numbers they provide also suffer (or is the setup different for those).

In any case wouldn't you or craig be able to set monitoring for this to see if this is indeed what's happening....
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