asterisk users - Aug 2003 - Great concept but a few issues unresolved

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?

 

 

The first issue I have noticed is with DMTF tones dialed from incoming
calls via iConnectHere.

 

NOTICE[18445]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 19
received

NOTICE[18445]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 19
received

 

 

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.

 

 

 

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.

 

Here is the log from the console when this happens:

 

-- Executing Macro("SIP/213.137.73.176:5060",
"dialpacket8|13057400221|70") in new stack

    -- Executing Dial("SIP/213.137.73.176:5060",
"SIP/13057400221@packet8.net|70") in new stack

    -- Called 13057400221@packet8.net

    -- SIP/packet8.net-f671 answered SIP/213.137.73.176:5060

    -- Attempting native bridge of SIP/213.137.73.176:5060 and
SIP/packet8.net-f671

    -- Got SIP response 404 "Not Found" back from 213.137.73.176

 

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.

 

 

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.

 

 

Thanks in advance for anyone that has even the slightest clue to what is
going on.

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