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���
C�sar Garc�a. Director de Sistemas, IdecNet S.A. Centro de Gesti�n de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - Espa�a. Tfn: +34 828 111 000 Ext: 340
Henry Jensen escribi�:
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/[EMAIL PROTECTED]/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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