[Asterisk-Dev] Problem In transfer call

Manoj K Gupta mgupta at spgsolutions.com
Thu Jul 31 06:42:50 MST 2003


Hi Michael,

Ok, i got that.But is call forwarding is also not supported?
Then can you suggest me an alternate way to achieve this through asterisk(if
possible).
ie i want to call a PSTN number through zapata card(T1 say) and channel bank
connected to * from one of openphone.
Then shift the current call to another alias registered on
gatekeeper(GNUGK).

Basically this is the same situation as described below.

Thanks..

Rgds
Manoj K Gupta



----- Original Message -----
From: "Michael Manousos" <manousos at inaccessnetworks.com>
To: <asterisk-dev at lists.digium.com>
Sent: Tuesday, July 29, 2003 3:42 PM
Subject: Re: [Asterisk-Dev] Problem In transfer call


>
> The H.450.x series of services is not supported by asterisk-oh323.
> The transfer key on the OpenPhone is trying to do it with one
> of these services. I guess that something is not handled
> right in asterisk-oh323, so you get the segfault.
>
> I'll try to fix it (the segfault) and let you know.
>
> Michael.
>
>
>
> Manoj K Gupta wrote:
> > Hi list,
> >
> > I am noticing a following error while trying the following scenerio.
> > I am using inaccess networks' h323channel driver version 0.5.4 with the
> > 25 july code of asterisk from cvs.
> >
> > I used two openphone from two machines in a lan environment with h323
> > alias as 001 and 002 respt to register with GNUGK on that lan only.
> >
> > And  asterisk is registered with GNUGK as exten 005
> >
> > ie
> >
> > `asterisk
> >     +           ------------OpenPhone  001
> > GNUGK    ________OpenPHone 002
> >
> > I first make a call using 001 to asterisk. at 005 using g.723(My
> > asterisk conf is having g.723 for test) and then use transfer button on
> > openphone to transfer  the current call to 002 (ie other openphone).The
> > call was transferrred but as soon as i recieved the call on 002, the
> > call was discconnected with the following error messages.
> >
> >
> >
> > WARNING[319513]: File chan_oh323.c, Line 2154 (alerted_h323_connection):
> > Call with reference 25538 in unexpected state (4).
> >
> > WARNING[286741]: File chan_oh323.c, Line 966 (oh323_read): H323:25538:
> > Invalid size for G.723.1 (24 bytes).
> >
> > WARNING[286741]: File chan_oh323.c, Line 966 (oh323_read): H323:25538:
> > Invalid size for G.723.1 (1 bytes).
> >
> > 3:38.777 H225 Caller:8146688 H225 Received connect PDU.
> >
> > WARNING[286741]: File frame.c, Line 76 (ast_smoother_feed): Smoother was
> > working on 1108521436 format frames, now
> >
> > trying to feed 1?
> >
> > ERROR[286741]: File chan_oh323.c, Line 1121 (oh323_write): H323:25538:
> > Failed to fill smoother.
> >
> > WARNING[286741]: File file.c, Line 509 (ast_readaudio_callback): Failed
> > to write frame
> >
> > == Spawn extension (default, s, 5) exited non-zero on 'H323:25538'
> >
> > -- Hungup 'H323:25538'
> >
> > 3:38.858 LogChanTx:814b900 Codec Audio read failed:
> >
> > 3:38.863 H225 Caller:8146688 H225 Read error (0):
> >
> > I tried using the other codec like GSM , but the same result.But when i
> > used G.711 from 001 to 005(Asterisk) then the call was able to last for
> > 1-2 minutes and then the asterisk crashed.
> >
> > I would be thankful if somebody can explain the reason or the soultion.I
> > suspect if asterisk-oh323 supports this kind of TRANSFER or there is
> > some probloem with asterisk itself.
> >
> >
> >
>
>
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>
>




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