[Asterisk-video] how to setup SIP-324M gateway

[Arnold P. Siboro]

I did not install the AMR codec, but then there was no codec related
error message. I then did instll AMR by patching and recompiling
Asterisk as instructed on this URL
http://lists.digium.com/pipermail/asterisk-dev/2007-March/026446.html
And I believe it's put in place, since Asterisk log shows as follows:

logger.c:   == Registered translator 'amrtolin' from format amr to slin,
cost 2
logger.c:   == Registered translator 'lintoamr' from format slin to amr,
cost 11
logger.c: codec_amr.so => (AMR Coder/Decoder)

With this dialplan:

[from-zaptel]
exten => s,1,h324m_gw(s at threegvideo)

[threegvideo]
exten => s,1,Dial(SIP/1002)

I could finally got the call on X-lite (X-Lite rings) but when it is
picked up the call was disconnected.

    -- Accepting data call from '08017734983' to 's' on channel 0/1, span 1
    -- Executing [s at from-zaptel:1] h324m_gw("Zap/1-1", "s at threegvideo") in new stack
    -- Executing [s at threegvideo:1] Dial("Local/s at threegvideo-6e38,2", "SIP/1002") in new stack
    -- Called 1002
    -- SIP/1002-08d9ee60 is ringing
    -- SIP/1002-08d9ee60 answered Local/s at threegvideo-6e38,2
  == Spawn extension (threegvideo, s, 1) exited non-zero on 'Local/s at threegvideo-6e38,2'

I wonder how to find out what is the reason of disconnection (are there
ways to investigate via logs for example)? X-Lite does not support AMR 
(Supported audio codecs are Broadvoice-32, Broadvoice-32 FEC, G.711aLaw,
G.711uLaw, GSM, iLBC, Speex), but Asterisk is supposed to handle
transcoding between AMR and for example G.711, isn't?


Pada Mon, 23 Jul 2007 13:54:54 +0200
si Klaus Darilion <klaus.mailinglists at pernau.at> bilang:

> Have you installed the AMR codec?
> 
> regards
> klaus
> 
> Arnold P. Siboro wrote:
> > Actually the instruction does not mention the Answer line, so I fixed
> > the configuration to as follows:
> > 
> > [from-zaptel]
> > exten => s,1,h324m_gw(s at threegvideo)
> > 
> > [threegvideo]
> > exten => s,1,Dial(SIP/1002)
> > 
> > But the caller keeps ringing while callee (1002) receives nothing, as
> > follows:
> > 
> > Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on asterisk1 (pid = 22790)
> > ionerisk1*CLI>
> > Verbosity is at least 3
> >     -- Going to extension s|1 because of immediate=yes
> >     -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
> >     -- Executing [s at from-zaptel:1] h324m_gw("Zap/1-1", "s at threegvideo") in new stack
> >     -- Executing [s at threegvideo:1] Dial("Local/s at threegvideo-3a33,2", "SIP/1002") in new stack
> >     -- Couldn't call 1002
> >   == Everyone is busy/congested at this time (0:0/0/0)
> >   == Auto fallthrough, channel 'Local/s at threegvideo-3a33,2' status is 'CHANUNAVAIL'
> >   == Spawn extension (from-zaptel, s, 1) exited non-zero on 'Zap/1-1'
> >     -- Hungup 'Zap/1-1'
> > 
> > 
> > Pada Mon, 23 Jul 2007 14:54:08 +0900
> > si "Arnold P. Siboro" <asiboro at maltech.jp> bilang:
> > 
> >> I got my Asterisk box running and tested with ISDN line. Furthermore, h324m_loopback()
> >> also worked perfectly. I want to setup a SIP-324M gateway, following the
> >> instruction on libh324m gateway, I set it as follows:
> >>
> >> [from-zaptel]
> >> exten => _.,1,Answer
> >> ;exten => s,10,h324m_gw(SIP/1002)
> >> exten => _X.,1,h324m_gw(s at threegvideo)
> >>
> >> [threegvideo]
> >> exten => s,1,Dial(SIP/1002)
> >>
> >> However, it does not work (caller keeps ringing but callee does not get
> >> does not response), giving the following message:
> >>
> >>
> >> Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on asterisk1 (pid = 22790)
> >> ionerisk1*CLI>
> >> Verbosity is at least 3
> >>   == Parsing '/etc/asterisk/manager.conf': Found
> >>   == Parsing '/etc/asterisk/manager_custom.conf': Found
> >>   == Manager 'admin' logged on from 127.0.0.1
> >>     -- Going to extension s|1 because of immediate=yes
> >>     -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
> >>     -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
> >>   == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
> >>     -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
> >>     -- Hungup 'Zap/1-1'
> >>
> >> Is it codec problem? I was kind of expecting some message if it's about
> >> codec. BTW, on the SIP end I am using X-Lite, which I think does not
> >> have the right audio codec to talk with h324m endpoint.




Arnold P. Siboro (asiboro at maltech.jp)

"Anyone who has never made a mistake has never tried anything new." 
                                          --  Albert Einstein