[Asterisk-Users] atxfer features in stable release.

[Cesar Garcia]

Hi all.

At the end, i get atxfer with sip dowloading head cvs version of 
asterisk and this is ok, but now i have errors with h323.

following the instructions i could compile h323 channel and load it, but 
when i call from sip to h323 or viceversa, i obtain this.

debug
-------------
May  4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 4/4)
May  4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 4/4)
May  4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 4/4)
May  4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 4/4)
     -- H323/as5300-1.lpa.idec.net answered SIP/u0001-fbca
May  4 12:12:07 WARNING[14186]: channel.c:2261 
ast_channel_make_compatible: No path to translate from SIP/u0001-fbca(4) 
to H323/212.xxx.xxx.xxx(256)
May  4 12:12:07 WARNING[14186]: app_dial.c:1315 dial_exec_full: Had to 
drop call because I couldn't make SIP/u0001-fbca compatible with 
H323/212.xxx.xxx.xxx
   == Spawn extension (default, 828111044, 1) exited non-zero on 
'SIP/u0001-fbca'
-------------
end debug


in the stable version, all its ok....

WHEN ATXFER AND THE REST OF FEATURESMAP FEATURES IN THE STABLE RELEASE?


Best Regards¡¡¡


César García.
    Director de Sistemas, IdecNet S.A.
    Centro de Gestión de Red.
    Edificio IdecNet. C/Juan XXIII 44.
    E-35004, Las Palmas de Gran Canaria,
    Islas Canarias - España.
    Tfn:  +34 828 111 000 Ext: 340

Henry Jensen escribió:
> Hello,
> 
> I have 2 *, one is between a Siemens HiPath and  the PSTN, having two PRIs
> connected to each side.
> 
> When I call the Hipath to administer it (with Siemens HiPath Manager), I
> usually call through the PSTN and all wents well.
> 
> However, I have a second Asterisk and when I call the first Asterisk trough
> the second to connect to the HiPath, the call comes not through.
> 
> To show you what I mean:
> 
> This works:
> 
> HiPath Manager -- ISDN PBX -- PSTN -- Asterisk1 -- HiPath
> 
> 
> This doesn't work:
> 
> HiPath Manager -- ISDN PBX -- Asterisk2 -- Asterisk1 -- HiPath
> 
> 
> Note: Voice calls are working perfectly, it's only the data calls that
> doesn't work.
> 
> 
> The debug output shows the following:
> 
> ----------------------------------------------------------------------------
>  -- Accepting unauthenticated call from XXX.XXX.XX.XX, requested format =
> 8, actual format = 8
> -- Executing Dial("IAX2/xxxxxx at xxxxxxxx/1", "Zap/g1/12345678") in
> new stack
> -- Called g1/12345678
> -- Executing Dial("Zap/5-1", "Zap/g2/12345678") in new stack
> -- Making new call for cr 32776
> 
>>Protocol Discriminator: Q.931 (8)  len=39
>>Call Ref: len= 2 (reference 8/0x8) (Originator)
>>Message type: SETUP (5)
>>[04 03 80 90 a3]
>>Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info
>>transfer capability: Speech (0)
>>                            Ext: 1  Trans mode/rate:
>>                             64kbps, circuit-mode (16)
>>                             Ext: 1  User information
>>			layer 1: A-Law (35)
> 
> 
> 
> [...]
> 
>        -- Channel 0/1, span 2 got hangup
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate
> Disconnect Request
> 	
> ----------------------------------------------------------------------------
> 
> I think the problem is the "transfer capability: Speech" line. It must be
> "transfer capability: Unrestricted digital information". 
> 
> Is there a way to set the transfer capability? I noticed there is a file
> app_settransfercapability.c in CVS (but not in 1.0.7).
> 
> Is this possible with IAX at all?
> 
> 
> Regards,
> Henry
> 
> 
> 
> 
> 
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