Hi folks,
[note added just before sending: You know how, by the time you
explicitly write down what's snagging you up, you figure out the
problem? Here's an example of that. I hope it will help someone else
either currently on the list, or checking the archives some day.]
[So the spots labelled QUESTION below become a bit rhetorical but feel
free to enlighten a newbie with any comments or advice.]
Just to give you some context on where I'm coming from with my
questions... I've been reading the list for about 4 or 6 months now, in
preparation to give Asterisk a shot. I'm very familiar with using
various Linuxes, mostly Redhat 7, 8, 9, ..., Fedora 6, and a bit of
Gentoo lately on various desktop and server machines around the home and
office and our company's data center rack for, I don't know, 6, 7
years. I have no problem building various packages, kernels, applying
patches, etc, so I'm not newbie-ish at all except now with Asterisk.
- I installed Ubuntu 7.04 on some modest old hardware (P3 550), (xubuntu
actually so it'd be a more modest Xfce-based X session, and I'll
probably disable X too after a while)
- I originally 'apt-get install'-ed asterisk which looked to work fine
but I noticed it was Asterisk 1.2
- following advice of www.the-asterisk-book.com, I 'apt-get remove'-d
that and installed build-essential and the other packages they mention
to build the latest Asterisk from source. There goes the simplicity but
what the heck.
- I followed Appendix A (to compile/install) and it worked exactly as
spelled out there... basically tar, configure, make, make install for
both zaptel and asterisk (with a 'modprobe ztdummy' after building
zaptel and before doing asterisk), make samples, make config (for the
startup scripts), and added ztdummy to /etc/modules
- I followed Chapter 1 (to configure a rudimentary 2 extension
system)... basically that had me replace sip.conf, extensions.conf, and
voicemail.conf with
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
secret=1234
host=dynamic
extensions.conf:
[others]
[my-phones]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,VoiceMail(2000,u)
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,VoiceMail(2001,u)
exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)
voicemail.conf:
[general]
format = wav
[default]
2000 => 4711,Martin1,[EMAIL PROTECTED]
2001 => 0815,Martin2,[EMAIL PROTECTED]
- I started asterisk with 'asterisk -vvvvvc' and it all looked pretty good
- now a bit of context... the above is all on this old machine, on a 2
point LAN, 192.168.26.2 and the desktop machine I'm testing it from is
192.168.26.1 (the desktop has 2 network cards, the other being
10.10.107.100 connected to LAN, router, rest of house, and internet)
- on the desktop machine, I set up Ekiga soft phone (and an Ekiga
account), and test it out two ways:
- the local echo test under the Test Settings button at the Audio
Devices step of the Druid, which sounds great
- a SIP call to their echo test (sip:[EMAIL PROTECTED]), it works okay...
the sound quality is a little choppy but at least I know my microphone,
headphones, soundcard, mixer settings and all that is more or less okay,
there is sound getting through
- now I set up Ekiga with the Asterisk details to try to call an
extension on the Asterisk machine, I set this in the Ekiga Account
Information dialog as follows:
Account Name: martin1 (I think this is just a label in the GUI, that
goes nowhere in terms of the protocol)
Registrar: 192.168.26.2
User: 2000 (as in my sip.conf)
Password: 1234 (as in my sip.conf)
[under More Options (it filled this part automatically without my typing
in here)]
Authenication Login: 2000
Realm/Domain: 192.168.26.2
Registration Timeout: 3600
- now I click the check box in Ekiga's Accounts window to un-register
the ekiga.net account and register the 192... account, I see in the
Asterisk terminal window:
*CLI> -- Registered SIP '2000' at 192.168.26.1 port 5061 expires 3600
-- Saved useragent "Ekiga/2.0.3" for peer 2000
which looks like a good and proper thing to see.
QUESTION 1: Can anyone confirm that the above is what I should see
and/or should I be seeing anything else more or less when I click
Register in Ekiga?
- I make a call to sip:[EMAIL PROTECTED] (the other extension in my
setup, for which there are no phones registered yet) and I see:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/2000-081ef708",
"SIP/2001|20") in new stack
[Oct 5 21:35:24] WARNING[1014]: app_dial.c:1111 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] VoiceMail("SIP/2000-081ef708",
"2001|u") in new stack
-- <SIP/2000-081ef708> Playing 'vm-theperson' (language 'en')
-- <SIP/2000-081ef708> Playing 'digits/2' (language 'en')
-- <SIP/2000-081ef708> Playing 'digits/0' (language 'en')
-- <SIP/2000-081ef708> Playing 'digits/0' (language 'en')
-- <SIP/2000-081ef708> Playing 'digits/1' (language 'en')
-- <SIP/2000-081ef708> Playing 'vm-isunavail' (language 'en')
-- <SIP/2000-081ef708> Playing 'vm-intro' (language 'en')
-- <SIP/2000-081ef708> Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/2001/tmp/z9a90d format: wav, 0x81f3388
but while it thinks it is "playing" all these voice mail prompts, I hear
silence.
QUESTION 2: is there some step I'm missing before I can expect those
voice mail prompts to be audible?
[actually going back after finding the solution and re-testing the
problem I noticed it wasn't actually silence, it was a short clipping
type noise and then silence]
- then I click Ekiga's hang up button and I see (again this seems like a
logical thing to see):
-- User hung up
== Spawn extension (my-phones, 2001, 2) exited non-zero on
'SIP/2000-081ef708'
- strangely (since it seems the voice mail isn't working), I can scp the
.wav files from the Asterisk machine's
/var/spool/asterisk/voicemail/default/2001/INBOX to the desktop machine
and play them in an audio player (they too sound choppy like the Ekiga
echo test but at least I can tell they are me talking, whatever I was
saying while I was on the call)
- the resolution was when I kind of remembered that the Ekiga echo test
it said Out: PCMU/H261 and same for In: but in my Asterisk test it said
GSM/ for both In and Out. So I started poking around with Ekiga's
Preferences > Codecs > Audio Codecs, which had all checked except the
last 3:
Speex (20.8 kbps)
GSM (13.2)
MS-GSM (13.0)
Speex (8.0)
PCMU (64)
PCMA (64)
G726-16k (16)
G721 (32)
LPC (2.5)
- I see that the set that are checked will change depending on whether
you chose LAN, Cable/DSL, or 56k modem in the Ekiga druid
- I disabled GSM, and on the next Asterisk test call, I heard all the
voice mail prompts that the CLI said it was playing (and Ekiga said
PCMU for both In and Out)
This is where the Yay (it's working) and Argh (that's all there was to
it!?) of the subject line comes in.
So anyway, thanks everyone for "listening" to me help myself, and for
good Qs and As earlier that probably indirectly steered me towards
figuring this out.
Cheers.
Martin
- also note, not related to my problem, but probably another newbie
gotcha, is that with this type of 2 point LAN behind another LAN (with
the desktop test machine being the router in between), the Ekiga echo
test to their server only works if I set the Network Settings > Listen
On: to be the upstream/internet NIC (10.10.107.100), and NAT Traversal:
STUN and STUN Server: stun.ekiga.net. Whereas the Asterisk tests only
work with Listen On: set to the other NIC (192.168.26.1), no NAT, no
STUN server.
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