Sipphone problems

It might be, or it could be something at your end. Might you have two phones or ata's behind one NAT box? If so, things might be a bit crunchy. Incoming packets for port 5060 (sip) and/or 5004 (rtp) could be going to the wrong phone/ATA when an externally initiated connection was starting up.

The only way to really tell what is going on is slap a tcpdump on the external line between the modem and nat-box and watch the transactions. From the SIP headers, it should be very obvious if your side is blowing off the incoming connections, or if the problem is further upstream.

Having just gone through this with two of my voip/pstn gatewaying providers, I can say that problems like this do exist. In my case, the a lack of any SIP packet coming in around the time the remote end got a fast busy is pretty strong proof that something upstream is either overloaded or misconfigured.

-wolfgang

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.