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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The past week or so I have been experimenting with Asterisk
and overall find it to be a nice software suite, although I have encountered
some problems, and have found almost no documentation (For example in sip.conf
I needed the commands fromuser= and fromdomain= and only figured out this was possible
after spending a few hours browsing on the internet and reviewing some person’s
configuration files they have posted). Is there at least a document that
explains all the possible config values and gives a sentence or two about their
use?</span></font></p>
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font-family:Arial'> </span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The first issue I have noticed is with DMTF tones dialed
from incoming calls via iConnectHere.</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>NOTICE[18445]: File rtp.c, Line 291 (ast_rtp_read): Unknown
RTP codec 19 received</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>NOTICE[18445]: File rtp.c, Line 291 (ast_rtp_read): Unknown
RTP codec 19 received</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>It will either double some digits or drop them (if I dial 16
it will instead dial 11 or 116 and get to the wrong extension). Is there
anything I can do to correct this issue? Currently I am using the eStara softphone
and it works great with dialing digits, so this seems to be mainly an issue for
incoming callers, same thing goes if I try to check voice messages, I must dial
each digit, wait a second and then dial the next one.</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The second issue is when I try to bridge incoming and
outgoing call (an external caller dials an extension which in turn is transferred
to a call dialed externally). The outgoing leg of the call cannot hear
anything, but the incoming leg of the call can. There is no NAT in this
situation, the Asterisk machine is connected directly to a switch which is
connected to a Cisco router which does no filtering or NAT, the machine has a
direct public connection to the internet. The only possible issue (and this
would be a rather odd one) is that the forward DNS and hostname of the machine
differs from the reverse DNS. I have attempted to use both Packet8 and iConnectHere
for these outgoing calls and they both yield different results.</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Here is the log from the console when this happens:</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>-- Executing Macro("SIP/213.137.73.176:5060",
"dialpacket8|13057400221|70") in new stack</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> -- Executing Dial("SIP/213.137.73.176:5060",
"SIP/13057400221@packet8.net|70") in new stack</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> -- Called 13057400221@packet8.net</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> -- SIP/packet8.net-f671 answered
SIP/213.137.73.176:5060</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> -- Attempting native bridge of
SIP/213.137.73.176:5060 and SIP/packet8.net-f671</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> -- Got SIP response 404 "Not
Found" back from 213.137.73.176</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>SIP/213.173.73.176 is the iConnectHere incoming connection.
I fail to understand why the originating server for the incoming call says that
something was not found…</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Is there any documentation I can read? I have yet to find
anything rather detailed on the Asterisk site. I have been having some other
issues with NAT and I assume that there must be an FAQ or technical document
somewhere that would cover basic use of SIP + * with NAT/PAT.</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks in advance for anyone that has even the slightest clue
to what is going on.</span></font></p>
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