[Asterisk-Users] Toll Free dialing problems

[Stewart Nelson]

> I've tried using iaxtel and BroadVoice to route toll free calls and the
> call appears to connect ok (see log snippet below) but it just rings and
> rings and eventually it times out and I get

> "The person you are calling is unavailable...."

Hi Shadow,

This is a common problem, not limited to VoIP.

Large toll-free users, such as IBM in your example, have enough clout
with their carrier, that they don't pay for minutes during the IVR
portion of the call.  This is accomplished by not sending answer
supervision until the call is sent to a human.  If you have SS7,
there is enough information to make this work properly.  If you
have a dumb POTS line, there is also no problem, because the CO
switch takes care of it for you.  But with an interface of moderate
intelligence, such as T1, or sometimes with PRI, SIP, or H.323, there
can be trouble.

First, verify that this is your problem.  Try calling (888) 746-7777
via Broadvoice.  You should hear the "call the talk line" advertisement.
Or, call a toll-free number that is answered by a human; it should
work ok.

If you have trouble with *all* toll-free numbers, see if setting
pedantic=yes in sip.conf helps (using CVS HEAD).  If not, post a
more detailed log.

However, if your problem is as described above, use Ethereal to
capture and play some audio from BroadVoice during the 183 Progress.
If you hear ringing, the problem is at BroadVoice and you'll have
to get them to fix it, or find another provider.  But if you hear
IBM's IVR, then either Asterisk is not passing the audio properly
to your client (IMO unlikely, use Ethereal to check), or your
client is not processing the Progress correctly (test with a
different SIP client or a non-VoIP extension).  Once you can
hear the IVR, you may have trouble getting outbound DTMF to
work during Progress.  Your phone or ATA may have an option
"send RTP during Progress" or something similar.

Good luck,

Stewart

> Log snippet below:

    -- Executing Dial("SIP/116-3e81",
"SIP/18887467426 at sip.broadvoice.com|45") in new stack
    -- Called 18887467426 at sip.broadvoice.com
    -- SIP/sip.broadvoice.com-ace8 is making progress passing it to
SIP/116-3e81
    -- Nobody picked up in 45000 ms
    -- Executing Congestion("SIP/116-3e81", "") in new stack
  == Spawn extension (inside, 18887467426, 2) exited non-zero on
'SIP/116-3e81'