I have a Sipura 3000 with my PSTN line connected to the FXO port. I have the Sipura 3000 set to “0 second PSTN Answer Delay” so that any incoming calls are routed directly to Asterisk.
However, when somebody calls, they get a stuttered ring. This means that they will hear an initial ring, but then it is cut-off by the Asterisk ring (eg. dialing an extension).
Can I elimate this first ring, or is it coming from the PSTN line? (eg. I have no control over it) I don’t think VoIP incoming lines have this problem, it’s only PSTN lines.
After doing some searching I’ve found “phones.conf” with an option called mode. What does this do? The possibly values are “dialtone”, “intermediate”, “fxo” and “fxs”. Do I need to change any of these to fix my problem?
Here’s my understanding. The sound your caller hears is called ringback, and it is not necessarily timed with your actual incoming ring. The ringback, therefore, may start before your phone starts to ring. I don’t think there’s anything you can do about it.
You might plug a regular phone directly into your pstn line and listen to see how long it rings before the Sipura box actually answers. It could have a bit of lag, I suppose.
Also, you could forward your PSTN line to a VoIP DID so there is never any ringing.
Another thought… It sounds like you have Asterisk answer and then immediately dial an extension, right? Why not just leave the line unanswered but dial the extension anyway, and let the PSTN line get taken off hook only when/if you (or voicemail) answers the extension?
I have Asterisk Answer() and then Dial() so that I can monitor the line for DTMF tones. This is done so that I can call in, press # and enter a “secret menu” for listening to my voicemail, getting the weather forecast, etc.
I might consider turning it off and only answering it right before the voicemail picks up. This would be an inconvience to me (since I would need to wait five to six rings before I could listen to my voicemail), but it would eliminate the “stuttered ring”.
I’d also like to forward my PSTN line to my VOIP line, but the entire point of my project is to save money and line forwarding costs extra money… (or is there another way I don’t know about aside from paying Verizon?)
Do you normally accept incoming calls on your VoIP line (via a DID) or is it for termination only? If you have a DID you’re not using, that could be your “back line”.
I’m in the process of getting a DID and will perhaps port my local number over to my VOIP provider (EXGN), but for now I intend to have two DIDs (one from EXGN and one from my local PSTN).
Can you explain what you mean by using it as my back line?
I just meant that you could have a private number you use just to call in get to your special menu. Since you talked about calls coming in to your PSTN line, but not having call forwarding, I thought you might have an unused DID from a VoIP provider that could be used for this.
If you ever do port your home number and go exclusively VoIP, you won’t have your ringing problem, but I’m not sure if * can monitor for dialed digits while an incoming VoIP call is still ringing. Anyone?