[asterisk-dev] RTP Audio Problem, No Audio Passed

[Nicholas Blasgen]

Sorry guys, seems I'm a dumb-ass.  I love it how after writing these long
e-mails with all the descriptions of how things interact, I figure out the
problem based on my words alone.  I copied the configuration files from a
system that had an internal IP address (192.168.X.X) to one that had an
external IP.  Only way to get an internal IP address Asterisk box to work is
to set the localnet and external options in sip.conf.  So all these problems
were because the Asterisk box thought its IP address was different than its
real address.

While this issue was all mine, I'm surprised that Asterisk didn't notice
that its IP address wasn't even within the "LOCALNET" IP block and therefore
couldn't transmit to the local range.  If Asterisk's IP isn't in the local
range, it should at least give an error for that if not ignore the setting
for the external IP.  I'm always surprised by applications that allow you to
set a data path (route) which you can't reach with your current IP and
Netmask settings.  I expect there is a reason for this, but as a network
tech I haven't figure out one.

Thanks anyways guys.  This list at least allows me to dwell better on the
Asterisk issues I have.  By talking things out I tend to figure out the
problem.  5 hours of work and I get no where, I send an e-mail to this list
and 5 minutes later I've figured it out.  Have a nice night!

/Nicholas Blasgen


On 10/2/07, Nicholas Blasgen <nicholas at blasgen.com> wrote:
>
> I've got two Asterisk boxes.  One has been designed as a copy-cat of the
> main production one.  I've simply installed Asterisk and copied over all the
> configuration files from the working production server to a newly setup
> server.  They're both using ViaTalk SIP trunks for outgoing calls, though
> they have different numbers they're using.  Production server, which works
> just fine, is sitting in the DMZ of an office NAT.  New server is a
> dedicated colocation with it's own IP address.  If you can't tell, I'm
> having issues with the new system.  The thing I'm trying to fix, which I
> think will fix any other issues, is that I can't hear the "ring" audio on
> the calls.  I can't hear any audio period (niether party can hear eachother
> once connected), but I'd expect to hear the "ring" sound in any situation.
>
> If I write a call file to call one phone number and execute the Dial
> application to call another, both systems have no issue.  But if I dial a
> "Local" channel and use that to call the first number, it works just great
> on the old box but doesn't work on the new box.
>
> GOOD BOX: Asterisk SVN-branch-1.4-r81291
> BAD BOX: Asterisk SVN-branch-1.4-r83348
>
> ======== CALL FILE ==========
> Channel: Local/nick at realtimedb
> MaxRetries: 0
> RetryTime: 15
> WaitTime: 30
> Application: Macro
> Data: which-line|4084979796
> ===========================
>
> ====== MINOR VERBOSE ======
> ======== GOOD BOX =========
>     -- Executing Dial("Local/nick at realtimedb-fe60,2", "
> Local/4083951234 at outgoing |20")
>     -- Called 4083951234 at outgoing
> [ . . . ]
>     -- Executing [s at macro-which-line:9] Dial("
> Local/4083951234 at outgoing-5d4c,2", "SIP/trunk0/14083951234") in new stack
>     -- Called trunk0/14083951234
>     -- SIP/trunk0-09dabf40 is making progress passing it to
> Local/4083951234 at outgoing-5d4c,2
>     -- Local/4083951234 at outgoing-5d4c,1 is making progress passing it to
> Local/nick at realtimedb-fe60,2
>     -- SIP/trunk0-09dabf40 answered Local/4083951234 at outgoing-5d4c,2
>     -- Local/4083951234 at outgoing-5d4c,1 answered
> Local/nick at realtimedb-fe60,2
> [ . . . ]
>     -- Executing [s at macro-which-line:9] Dial("Local/nick at realtimedb-fe60,1",
> "SIP/trunk0/14084979796") in new stack
>     -- Called trunk0/14084979796
>   == Spawn extension (macro-which-line, s, 9) exited non-zero on '
> Local/4083951234 at outgoing-5d4c,2' in macro 'which-line'
>   == Spawn extension (macro-which-line, s, 9) exited non-zero on '
> Local/4083951234 at outgoing-5d4c,2'
>   == Spawn extension (realtimedb, nick, 2) exited non-zero on '
> Local/nick at realtimedb-fe60,2'
>     -- SIP/trunk0-09d8f2c8 is making progress passing it to
> SIP/trunk0-09dabf40
> ==========================
> ======== BADD BOX =========
>     -- Executing Dial("Local/nick at realtimedb-7bf7,2", "
> Local/4083951234 at outgoing|20")
>     -- Called 4083951234 at outgoing
> [Oct  2 22:26:37] NOTICE[2721]: cdr.c:434 ast_cdr_free: CDR on channel
> 'SIP/trunk0-08f9f610' not posted
> [ . . . ]
>     -- Executing [s at macro-which-line:9] Dial("
> Local/4083951234 at outgoing-bb40,2", "SIP/trunk0/14083951234") in new stack
>     -- Called trunk0/14083951234
>     -- SIP/trunk0-08f8bc88 is making progress passing it to
> Local/4083951234 at outgoing-bb40,2
>     -- Local/4083951234 at outgoing-bb40,1 is making progress passing it to
> Local/nick at realtimedb-7bf7 ,2
>     -- SIP/trunk0-08f8bc88 answered Local/4083951234 at outgoing-bb40,2
>     -- Local/4083951234 at outgoing-bb40,1 answered
> Local/nick at realtimedb-7bf7,2
> [Oct  2 22:26:42] WARNING[2719]: pbx.c:5141 ast_pbx_outgoing_app:
> Local/nick at realtimedb-7bf7,1 already has a call record??
> [Oct  2 22:26:42] NOTICE[2719]: cdr.c:434 ast_cdr_free: CDR on channel
> 'SIP/trunk0-08f9a950' not posted
> [ . . . ]
>     -- Executing [s at macro-which-line :9] Dial("Local/nick at realtimedb-7bf7,1",
> "SIP/trunk0/14084979796") in new stack
>     -- Called trunk0/14084979796
>     -- SIP/trunk0-08fa4960 is making progress passing it to
> Local/nick at realtimedb-7bf7 ,1
> ==========================
> ==========================
>
> The main thing to note from these is that in the case of the "good" box,
> you see it drop the "masq" names and when the call is getting patched it's
> refering to them only as making progress connecting trunk0-id1 to
> trunk0-id2.  In the case of the bad box, it's connecting trunk0-id1 to an
> extension that I guess isn't properly linked.
>
> When I turn on RTP debugging, the BAD system doesn't show anything
> ever.  The GOOD system shows plenty of RTP traffic while the call is being
> setup but nothing after it starts to ring.
>
> --
> /Nick
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