A small experiment with Ohphone - Asterisk - X-Lite

Inspired by a test found on the ooh323c-devel mailing list, we tried shortly to call a SIP extension of Asterisk by using Ohphone:

As a test first we start the gatekeeper gnugk:

gnugk -ttt

We then start GnomeMeeting on another linux workstation.

On a third linux workstation we dial

ohphone -g 129.125.71.171 -u 601 -e -f -T 602

this succeeds in calling the GnomeMeeting. see the log:

15:21:28 Started GnomeMeeting 1.2.3 for user fabio
15:28:58 Gatekeeper set to GNU Gatekeeper@129.125.71.171
16:31:55 Call from 601
16:32:03 Answering incoming call
16:32:03 Opened Intel ICH5 for recording with plugin ALSA
16:32:03 Opened codec MS-GSM{sw} for transmission
16:32:03 Connected with 601 using Open H323 Project OhPhone	1.4.5
16:32:03 Closed codec MS-GSM{sw} which was opened for transmission

On the other side, Ohphone reports:

OhPhone Version 1.4.5 by Open H323 Project on Unix Linux (2.6.18-3-686-i686)

Incoming channel port ranges 5000 to 5999
Local username: 601
TerminateOnHangup is 1
Auto answer is 0
DialAfterHangup is 0
FastStart is 0
H245Tunnelling is 0
SilenceSupression is 0
H245InSetup is 1
Jitter buffer: 50-250 ms
Connect port: 1720

Video receive disabled

Video transmit disabled


Sound output device: "Default"
Sound  input device: "Default"
User Input Send Mode: as H.245 string
Codecs (in preference order):
 Table:
   G.711-ALaw-64k <1>
   G.711-uLaw-64k <2>
   G.726-16k{sw} <3>
   G.726-24k{sw} <4>
   G.726-32k{sw} <5>
   G.726-40k{sw} <6>
   GSM-06.10{sw} <7>
   LPC-10{sw} <8>
   MS-GSM{sw} <9>
   MS-IMA-ADPCM{sw} <10>
   SpeexIETFNarrow-11k{sw} <11>
   SpeexIETFNarrow-15k{sw} <12>
   SpeexIETFNarrow-18.2k{sw} <13>
   SpeexIETFNarrow-24.6k{sw} <14>
   SpeexIETFNarrow-5.95k{sw} <15>
   SpeexIETFNarrow-8k{sw} <16>
   SpeexIETFWide-11.55k{sw} <17>
   SpeexIETFWide-17.6k{sw} <18>
   SpeexIETFWide-28.6k{sw} <19>
   SpeexNarrow-11k{sw} <20>
   SpeexNarrow-15k{sw} <21>
   SpeexNarrow-18.2k{sw} <22>
   SpeexNarrow-24.6k{sw} <23>
   SpeexNarrow-5.95k{sw} <24>
   SpeexNarrow-8k{sw} <25>
   SpeexWNarrow-8k{sw} <26>
   SpeexWide-11.55k{sw} <27>
   SpeexWide-17.6k{sw} <28>
   SpeexWide-28.6k{sw} <29>
   T.38-UDP <30>
   UserInput/hookflash <31>
   UserInput/basicString <32>
   UserInput/dtmf <33>
   UserInput/RFC2833 <34>
 Set:
   0:
     0:
       G.711-ALaw-64k <1>
       G.711-uLaw-64k <2>
       G.726-16k{sw} <3>
       G.726-24k{sw} <4>
       G.726-32k{sw} <5>
       G.726-40k{sw} <6>
       GSM-06.10{sw} <7>
       LPC-10{sw} <8>
       MS-GSM{sw} <9>
       MS-IMA-ADPCM{sw} <10>
       SpeexIETFNarrow-11k{sw} <11>
       SpeexIETFNarrow-15k{sw} <12>
       SpeexIETFNarrow-18.2k{sw} <13>
       SpeexIETFNarrow-24.6k{sw} <14>
       SpeexIETFNarrow-5.95k{sw} <15>
       SpeexIETFNarrow-8k{sw} <16>
       SpeexIETFWide-11.55k{sw} <17>
       SpeexIETFWide-17.6k{sw} <18>
       SpeexIETFWide-28.6k{sw} <19>
       SpeexNarrow-11k{sw} <20>
       SpeexNarrow-15k{sw} <21>
       SpeexNarrow-18.2k{sw} <22>
       SpeexNarrow-24.6k{sw} <23>
       SpeexNarrow-5.95k{sw} <24>
       SpeexNarrow-8k{sw} <25>
       SpeexWNarrow-8k{sw} <26>
       SpeexWide-11.55k{sw} <27>
       SpeexWide-17.6k{sw} <28>
       SpeexWide-28.6k{sw} <29>
     1:
       T.38-UDP <30>
     2:
       UserInput/hookflash <31>
     3:
       UserInput/basicString <32>
       UserInput/dtmf <33>
       UserInput/RFC2833 <34>


Listening interfaces : ALL:1720
Gatekeeper set: GNU Gatekeeper@129.125.71.171
601 is calling host 602
Command ? Ringing phone for "Fabio Bracci" ...
Call with "Fabio Bracci" established.
Could not open sound device Default - Check permissions or full duplex capability.
"Fabio Bracci" has cleared the call, duration  0:31

Command ? x
Exiting.
OhPhone ended.

We then try to call an extension on Asterisk:

fabio@wingtip112:~$ ohphone -g 129.125.71.171 -u 601 -e -f -T 502
OhPhone Version 1.4.5 by Open H323 Project on Unix Linux (2.6.18-3-686-i686)

Incoming channel port ranges 5000 to 5999
Local username: 601
TerminateOnHangup is 1
Auto answer is 0
DialAfterHangup is 0
FastStart is 0
H245Tunnelling is 0
SilenceSupression is 0
H245InSetup is 1
Jitter buffer: 50-250 ms
Connect port: 1720

Video receive disabled

Video transmit disabled


Sound output device: "Default"
Sound  input device: "Default"
User Input Send Mode: as H.245 string
Codecs (in preference order):
 Table:
   G.711-ALaw-64k <1>
   ...

Listening interfaces : ALL:1720
Gatekeeper set: GNU Gatekeeper@129.125.71.171
601 is calling host 502
Command ? Gatekeeper could not find user "502"

So now the problem seems to be that asterisk doesn't report his extensions to the gatekeeper. Further research is needed.