Sipphone problems

Discussion in 'VOIP' started by Vox Humana, Dec 27, 2005.

  1. Vox Humana

    Vox Humana Guest

    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.
    Vox Humana, Dec 27, 2005
    #1
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  2. "Vox Humana" <> writes:
    > 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
    Wolfgang S. Rupprecht, Dec 28, 2005
    #2
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  3. Vox Humana

    Vox Humana Guest

    "Wolfgang S. Rupprecht"
    <> wrote in
    message news:...
    >
    > "Vox Humana" <> writes:
    > > 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.
    Vox Humana, Dec 28, 2005
    #3
  4. "Vox Humana" <> writes:
    > 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
    Wolfgang S. Rupprecht, Dec 28, 2005
    #4
  5. Vox Humana

    Guest

    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
    , Dec 29, 2005
    #5
  6. Vox Humana

    Vox Humana Guest

    <> wrote in message
    news:...
    > 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.


    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
    2. 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.
    Vox Humana, Dec 30, 2005
    #6
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