We had this problem a few months ago but they resolved it for us. I really don't remember more than that.

Darren Wiebe
[EMAIL PROTECTED]

Tom Vile wrote:

I have been battling this problem for 2 months with no resolution as of yet with TelaSIP. I am told that it is a provider problem(Level 3) because all TelaSIP is doing is passing the call directly to them once the call comes through.

Anyone else having this issue with TelaSIP or Level3?

On 10/10/05, *John Millican* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:

    Hello all,
    yes there is a lot of information about this on the wiki and in
    past posts on
    this list but have not found anything that has solved my problem.
    setup is:
    phone-->PAP2-na-->asterisk v1.0.9(in house on local subnet dtmf is
    inband)--->PSTN--->Telisip---->asterisk box at colo v1.0.9 VoIP
    only.  I have
    only access to dial up so can not go VoIP out of the house.
    In extensions.conf  on colo * i have some logic that based on
    callerid lets me
    hit a single digit to get to DISA, this work every time.
    the problem is that when i enter a number for DISA to dial i get
    duplicate
    digits, example i enter 6037862111 and disa tries to dial
    6003778621.  I have
    tried setting relaxdtmf=yes in sip.conf with no luck.  I have read
    on the
    wiki that RFC2833 should work, but alas its a no go.  I am also
    using ulaw
    which should not be distorting the dtmf through compresion,
    correct? Also
    with RFC2833 it should not matter? Everything works great
    otherwise. sip.conf
    for colo * is posted below:
    [general]
    context=telasip
    port=5060
    bindaddr=0.0.0.0 <http://0.0.0.0>
    srvlookup=yes

    disallow=all                    ; First disallow all codecs
    allow=ulaw

    register => username:[EMAIL PROTECTED]
    <mailto:username:[EMAIL PROTECTED]>

    [telasip]
    type=peer
    username=*****
    fromuser=*****
    authname=*****
    secret=*****
    host=gw3.telasip.com <http://gw3.telasip.com>
    context=default
    dtmfmode=RFC2833
    disallow=all
    allow=ulaw
    canreinvite=no
    nat=no

    Thanks in advance for any help
    John Millican
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com <http://www.baldwintechsolutions.com>
Phone: 518-631-2855 x205
Fax:     518-631-2856

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