[Asterisk-Users] Really basic stuff :(

[Gavin Hamill]

Hi :)

I've had all this working before, but I'm revisiting it, and in short, I
currently have huge problems receiving incoming calls. I've been trying
with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel
and libpri as of yesterday afternoon.

Would someone mind helping? :)

My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set 
as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec
etc.) goes directly to that machine. I am not doing any firewalling, nor 
is my ISP.

I've made my configuration as superficial as I can to ease diagnosis:

root at eddie:/etc/asterisk# ls -l
-rw-r--r--    1 root     root          104 Jun 23 21:21 extensions.conf
-rw-r--r--    1 root     root          164 Jun 23 19:25 iax.conf
-rw-r--r--    1 root     root            0 Jun 22 15:36 modem.conf
-rw-r--r--    1 root     root          387 Jun 23 21:22 modules.conf
-rw-r--r--    1 root     root          363 Jun 23 21:19 sip.conf
-rw-r--r--    1 root     root            0 Jun 22 15:36 voicemail.conf

root at eddie:/etc/asterisk# more extensions.conf
[general]
static=no
writeprotect=yes

[default]
exten => 3333,1,Dial(IAX2/janie|20|tr)

root at eddie:/etc/asterisk# more iax.conf
[general]
port=5036

[janie]
type=friend
username=janie
secret=mysecret
host=dynamic
context=default
auth=md5
notransfer=yes

root at eddie:/etc/asterisk# more modules.conf
[modules]
autoload=yes
noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_skinny.so
noload => chan_mgcp.so
noload => chan_phone.so
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
noload => chan_zap.so

root at eddie:/etc/asterisk# more sip.conf
[general]
port = 5060
bindaddr = 10.0.0.1
context = default
disallow = all
allow = ulaw
allow = alaw
allow = gsm
externip = 213.232.83.29
localnet = 10.0.0.0
localmask = 255.255.255.0

register => 77830:MyPassword at fwd.pulver.com/3333

[fwd.pulver.com]
type=friend
secret=MyPassword
username=77830
host=fwd.pulver.com

===================================================

root at eddie:/etc/asterisk# asterisk -vvvvvvvvvvvc
[....]

Asterisk Ready.
*CLI> sip show registry
Host                  Username       Refresh State
192.246.69.223:5060   77830              120 Registered
*CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port     
Status
fwd.pulver.com/  192.246.69.223              255.255.255.255  5060     
Unmonitored

*CLI> iax2 show registry
Host                  Username    Perceived             Refresh  State
*CLI> iax2 show peers
Name/Username    Host            Mask             Port  Status
janie/janie      10.0.0.74  (D)  255.255.255.255  4569  Unmonitored

(janie is using iaxComm for Windows as the soft phone, and dialling 
'3333' from iaxComm causes a call to come in on 'line 2' of iaxComm)

If I now initiate an external call using FWD's "Call Me"

*CLI> ##### Testing 192.246.69.223 with 192.246.69.223
Target address 192.246.69.223 is not local, substituting externip
Setting NAT on RTP to -1
Stopping retransmission on '1710764988 at alphacp' of Response 1: Found

[15 seconds pass]

Auto destroying call '1710764988 at alphacp'


[20 seconds pass]


*CLI> ##### Testing 65.39.205.111 with 65.39.205.111
Target address 65.39.205.111 is not local, substituting externip
##### Testing 65.39.205.111 with 65.39.205.111
Target address 65.39.205.111 is not local, substituting externip
Check for res for
 is not a local user
build_route: Contact hop: sip:65.39.205.111:5060
    -- Executing Dial("SIP/fwd.pulver.com-0811c948", "IAX2/janie|20|tr") 
in new stack
SIMPLE DIAL (NO URL)
    -- Called janie
    -- Call accepted by 10.0.0.74 (format ULAW)
    -- Format for call is ULAW
    -- IAX2[janie]/4 is ringing

And so it is. I answer the softphone and:

Dropping incompatible voice frame on IAX2[janie]/4 of format GSM since 
our native format has changed to ULAW

screams up the screen for each frame...

Does this make sense to anyone? 

I have made a full SIP trace of the session available at 
http://gdh.ca/siptrace.txt if it helps! :)

Final note, I have tried 'nat=yes' and 'nat=no' in the
[fwd.pulver.com] section of sip.conf but it makes no difference :(

Cheers,
Gavin.