Sipphone problems

I subscribed to Sipphone about a month ago and bought a DID in another city (Cleveland) so my sister could call me toll-free. Incoming calls are free on Sipphone and she lives in the Cleveland metro area so the call is free for her.

The problem is that she rarely can complete a call. When she does, it takes several attempts. When she dials, my phone will ring once or twice. She hears a couple of rings and then either gets a message that says "All circuits are busy. Try again later," or she simply gets a fast busy signal. I rarely make calls from that line. The other line is configured for another voip service. Occasionally I will be unable to complete an outgoing call through Sipphone. I get a recording that says the call can not be completed as dialed.

I assume this is a Sipphone issue? I never have problems with my other VOIP provider.

It sure sounds like the setup is symmetrical between teliax and sipphone so if incoming works on one it should work on the other.

Any chance you can attach a hub (not switch) upstream of your d-link and attach a computer to this hub and watch the incoming packets? Or alternately can the d-link "router" be configured to log packet headers? Being able to tell if the SIP invites came in would be a very important data point.

I'm still looking for a good free per call DID too. For a while I was telling LD callers to call the free ipkall number and was only using the teliax DID for neighbors that objected to calling an LD number. The cost of in-state calls quickly passes the cost of out of state calls, so calling LD saves money at even commuting type distances. That helped to keep the number of calls to teliax relatively small and the per-minute fees never amounted to as much as the base fee for the DID. Unfortunately my ipkall call quality has taken a dive. I now get drop-outs of 3-5 seconds and have started telling people to only use that DID as a last resort.

-wolfgang

Thanks for the reply. I have a single ATA behind my D-Link router. The ATA has been assigned a static IP address and is in the DMZ. Line one is configured for one provider and is assigned to port 5060. The second line is configured for Sipphone and assigned to port 5061. I will take this up with the people at Sipphone. If I can't resolve the problem, I will just drop Sipphone. I would like to find a reliable provider with a pay-go plan that doesn't charge for incoming calls. I use Teliax as my primary VOIP provider and while their service is good, I hate the idea of paying for incoming calls. I rarely make outgoing calls. They charge 2 cents for the first 1- 60 seconds and 2 cents a minute thereafter for both incoming and outgoing calls.

I was thinking of using teliax for outgoing and SIPPhone for incoming. (Sipphone costs less for the DID and has free incoming, while Teliax has 0.03/minute for calls to Brazil.) Any ideas how to make this work, since you are running both? It looks like just having an ATA with 2 lines...what ATA are you using? Do you have 2 phones hooked up as well, or just one? I could see a benefit of having one phone for outgoing and one for incoming.

Many Thanks.

Jason

I have the Linksys PAP2-NA that was supplied by Teliax. Line 1 is configured for Teliax and Line 2 is configured for Sipphone. I have a combine-a-line plugged into both fx ports of the ATA. The combina-a-line is plugged into my home phone wiring. Unfortunately, the last line that was active is the default line. So if someone calls you on line1 and you want to call out on line 2, you have to go to the combine-a-line and select line

1. At some point I am going to replace all my phones with 2-line phones and get rid of the combine-a-line.

If you wanted, you could have a phone plugged into each fx port on the ATA, or you could plug the line you use most into your home's phone wiring and plug another phone into the other port. It all depends on the physical set-up you have.