I've been playing with SIP a lot lately and we rolled out some more of our
employees on SIP based long distance (testing to see if it's feasible to
collocate our Asterisk at our data center with a PRI - hence the talks
before about sharing PRI, etc.)..
I'm seeing an odd issue, I'm testing through IConnectHere with limited
success - however I seem to have one thing that may be reproducible (too
early to say if reproducible):
Layout: POTS based phone, hooked to Zhone Channel bank - Zhone
to Quad T1 Digium Card on Asterisk server. Asterisk then Dial/SIP to analog
based phone system.
Trying: Analog phone system (called party) says press 123 for sales.. We
press 123. Suddenly call quality is horrible (we can hear fine about 75% of
the time, other 25% no audio, called party says can't hear us hardly) -
other DTMF tones don't seem to go through right (maybe unclear so not being
recognized).
Works: Calling what I would guess is another digital PBX works fine using
this same layout. I've tried a few banks / large businesses and no
problems.
Works: Can call a phone via SIP and recognize DTMF from called party (i.e. #
to transfer)
Any ideas?
We should have another SIP based gateway up next week and we now have
pricing on DC based PRI that's good so we may just go with PRI. I'd rather
avoid the cross connect fees / loops on the PRI since we can get SIP based
easier and have plenty of bandwidth - it seems like just another thing to go
wrong in Asterisk using PRI instead of Cisco delivered SIP (maybe I'm
backwards on this).
PS: How does IAX handle DTMF between Asterisk servers?
Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers
BitShop, Inc. - <http://www.bitshop.com/> http://www.bitshop.com -
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I've been playing with SIP a lot lately and we rolled
out some more of our employees on SIP based long distance (testing to see if it's
feasible to collocate our Asterisk at our data center with a PRI - hence the
talks before about sharing PRI, etc.)..</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I'm seeing an odd issue, I'm testing through IConnectHere
with limited success - however I seem to have one thing that may be reproducible
(too early to say if reproducible):</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> Layout:
POTS based phone, hooked to Zhone Channel bank - Zhone to Quad T1 Digium
Card on Asterisk server. Asterisk then Dial/SIP to analog based phone
system. </span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Trying: Analog phone system (called
party) says press 123 for sales.. We press 123. Suddenly call quality is
horrible (we can hear fine about 75% of the time, other 25% no audio, called
party says can't hear us hardly) - other DTMF tones don't
seem to go through right (maybe unclear so not being recognized).</span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Works: Calling what I would guess is
another digital PBX works fine using this same layout. I've tried a
few banks / large businesses and no problems.</span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Works: Can call a phone via SIP and
recognize DTMF from called party (i.e. # to transfer)</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><br>
We should have another SIP based gateway up next week and we now have pricing
on DC based PRI that's good so we may just go with PRI. I'd
rather avoid the cross connect fees / loops on the PRI since we can get SIP
based easier and have plenty of bandwidth - it seems like just another
thing to go wrong in Asterisk using PRI instead of Cisco delivered SIP (maybe I'm
backwards on this).</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>PS: How does IAX handle DTMF between Asterisk servers?</span></font></p>
I've been playing with SIP a lot lately and we rolled out some more
of our employees on SIP based long distance (testing to see if it's
feasible to collocate our Asterisk at our data center with a PRI -
hence the talks before about sharing PRI, etc.)..
I'm seeing an odd issue, I'm testing through IConnectHere with
limited success - however I seem to have one thing that may be
reproducible (too early to say if reproducible):
Layout: POTS based phone, hooked to Zhone Channel bank
- Zhone to Quad T1 Digium Card on Asterisk server. Asterisk then
Dial/SIP to analog based phone system.
Trying: Analog phone system (called party) says press 123 for
sales.. We press 123. Suddenly call quality is horrible (we can
hear fine about 75% of the time, other 25% no audio, called party
says can't hear us hardly) - other DTMF tones don't seem to go
through right (maybe unclear so not being recognized).
No idea on this one. What is your dtmf setting on the SIP peer?
Quote:
Works: Calling what I would guess is another digital PBX works fine
using this same layout. I've tried a few banks / large businesses
and no problems.
Works: Can call a phone via SIP and recognize DTMF from called party
(i.e. # to transfer)
Still no "T" option working that I can determine, though. :( Not
that major an issue, but still problematic for when I want to send
someone to my IVR or send them to a DISA line after calling them.
Quote:
Any ideas?
Change dtmfmode= in your sip.conf file for that peer to either
rfc2833 or inband. I have yet to get iconnecthere working with DTMF,
and though some people here on the list say they've had it work, I
have not seen any configs, nor will they comment on how they did it.
Quote:
We should have another SIP based gateway up next week and we now
have pricing on DC based PRI that's good so we may just go with PRI.
I'd rather avoid the cross connect fees / loops on the PRI since we
can get SIP based easier and have plenty of bandwidth - it seems
like just another thing to go wrong in Asterisk using PRI instead of
Cisco delivered SIP (maybe I'm backwards on this).
I think I know the network configurations you're talking about. :)
By using a SIP provider, you lose some bandwidth advantages that IAX
can provide to you (if you want, or care.) However, you are correct
in that you don't have to pay for a PRI into your facility and then
connect it to your gear. If you can get SIP directly via IP (across
the Internet) is the "best" solution as long as
bandwidth/jitter/latency/cost of bits are acceptable for your
circumstances. The second best path would be to find a provider who
can co-locate an Asterisk box and give you zero-mile local loop costs
for a PRI into their switch; the cost for a 1u box plus some
bandwidth for IAX communications is almost always cheaper than the
delta of the cost for that same PRI being dragged through the RBOC.
Quote:
PS: How does IAX handle DTMF between Asterisk servers?
Works fine. I use it all the time. SIP -> * -> IAX -> * -> PRI
works like a charm, though the tones are a bit short by default.
I've never had problems with any of the IVR systems I've used, though.
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