voxlot
unfortunately, webcall does not work at all for me , i registered with my provider and i did enter my PSTN number by the choosing the porvider that should be calling that PSTN number as well as i entered the destination number with which provider to use (registered as well)
i have your call is being initiated, and still waiting for that ring. thanks |
Can you give us more detailed information please? For example, what provider(s) are you using, are both legs of Web Callback going to PSTN numbers, are both legs using the same provider, etc? Please provide as much information as possible.
Ron |
i have create two providers, one the voip provider i am using and the sacond on is sipphone.
your number: my PSTN number 1212xxxxxxxx voip provider1: provider i registered with voxlot to initiate the call from that account number to dial: 1747xxxxxxxx voip provider2: sipphone (i registered an account from sipphone with voxlot) when i click on "call", i should receive an inbound call dialed from the registered account of voip provider1 to my PSTN number and voxlot will be dialing 1747 (sipphone numnber) using voip provider 2 in this case sipphone and connect both calls together isnt it? on the other hand, is it compulsory that voip provider1 be one of your listed providers that you are peering with or it can be any provider, meaning not necessary to be one of your peering partners thnks |
If "Your Number" is '1212xxxxxxxx', you would have to specify a provider to the right of it that is capable of calling that PSTN number.
If "Number to Dial" is '1747xxxxxxxx', you would have to specify Sipphone or Sip Broker as the provider to the right of it. Sipphone numbers (area code 747) are not real PSTN numbers and can only be reached via Sipphone or Sip Broker as the provider. I just tried the above scenario (using both variations of "Number to Dial") and everything worked as expected. You should be able to use most any combination of providers on Web Callback assuming the number to the left of the selected provider is capable of being reached through that provider. Ron |
unfortunately, i tried all possible means and it does not work, what u explained is exactly what was entered and still nothing is happening.
do you want me to provide you with test numbers so you can test it yourself or possibly give you access to my login and check it out yourself thanks |
it is working from sipphone to FWD, i think the problem is my provider, if you wish i will provide you with a test account from my provider so you can test it as why it does not work
thanks |
i guess i know why, your server is asterisk right? if yes than you would need to do some amendment:
useragent=anything but not “asterisk” i guess u should use "voxalot" thanks |
Hi yaman,
Please try now with our recent change. |
yes, now it is working but after testing and hangup, i keep getting some calls couple of times to the the account created as the provider1, which i need to pick and hang up few times before that phone does not ring anymore
thanks a lot, i can provide this service to my clients, meaning i can post on my website for them to register with your beta and use it until your premium service become available thanks |
everything works fine, i was able to make two leg call using one account with one provider, meaning my number was called by account A with provider1 and the same account A with provider 1 called the destination number. the only problem i am having is:
while in conversation with the destination number the analogue phone that is connected to the ATA where the account A is configured keeps ringing until i finish my call and i need to pick th ephone and hang up couple of time before it stops ringing. any clues ?? thanks |
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