[asterisk-users] Problem with SIP

[Rodrigo Lang]

Hi, thanks a lot by the answers. But without the application Answer() the
problem remains.


Realized over a battery of tests and refined the problem. Follows:

A = External link that came with my Voip number.
B = Operator.
C = The extent to which A want to speak.

A called my number and B answer. If B try to transfer with blindxfer (#) to
C works fine. But if B try to transfer with atxfer (*2) he can talk to C,
only when B hangs up to complete the transfer begins to generate those
warnings on the cli. After the transfer using C atxfer not hear A, but A
hears C.

I believe it has become clearer now. And as he said, with any codec, and
only when the person connects to my VoIP trunks. I did the test with the
analogue trunks and atxfer worked normal.


Thanks,
Rodrigo Lang.



2010/7/20 Stefan Schmidt <sst at sil.at>

> Rodrigo Lang schrieb:
> > Good afternoon list.
> >
> > I'm experiencing a problem with my SIP channel's. When I have an
> > external connection for one of my SIP carrier's, I can listen to the
> > client and the client listens to me normally. The problem is when I
> > will transfer this connection, the call is mute for the extension I
> > have transfered. Only the client hears normally. In the console of
> > Asterisk generates the following warning:
> >
> > [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to
> > transmit frame type 64, while native formats is 0x2 (gsm) (2) read /
> > write = 0x40 (slin) (64) / 0x2 (gsm) (2)
> > [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to
> > transmit frame type 64, while native formats is 0x2 (gsm) (2) read /
> > write = 0x40 (slin) (64) / 0x2 (gsm) (2)
> >
> >
> > Detail, this happens with both the codec gsm, ulaw, alaw and g729 and
> > with any of my SIP carrier's (I own three). And only happens when the
> > call is transferred.
> >
> > Does anyone have any idea what could be?
> >
> > Thanks,
> > Rodrigo Lang.
> hello rodrigo,
>
> this is exactly the problem i had. Have a look at issue 17641
> (https://issues.asterisk.org/view.php?id=17641)
> There is a patch for asterisk 1.6.2.9 but its only a single row so you
> could easy find the position in app_dial.c to patch it by your own.
> the problem only occurs when you use answer in your dialplan. without an
> answer this wont happen.
>
>
> best regards.
>
> steve
>
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