Sipphone problems

Sipphone problems

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Subject Author Date
Sipphone problems Vox Humana 12-27-2005
---> Re: Sipphone problems Wolfgang S. Rup ..12-27-2005
Posted by Vox Humana on December 27, 2005, 6:04 pm
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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.



Posted by Wolfgang S. Rupprecht on December 27, 2005, 9:35 pm
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> I assume this is a Sipphone issue? I never have problems with my other VOIP
> provider.

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
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

Posted by Vox Humana on December 28, 2005, 3:17 pm
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"Wolfgang S. Rupprecht"
>
> > I assume this is a Sipphone issue? I never have problems with my other
VOIP
> > provider.
>
> 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.

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.



Posted by Wolfgang S. Rupprecht on December 28, 2005, 3:57 pm
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> 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.

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
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

Posted by on December 29, 2005, 1:15 pm
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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


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