[Asterisk-Users] Really basic stuff :(

[Senad Jordanovic]

Gavin Hamill wrote:
> 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.

I remember once having same issue like this using a softfone. Playing
with codecs on both sides did solve the problem.

Sorry, that I can not be specific since I just can not remember the
details.

Ta
Senad