[asterisk-users] Problem with SIP

[Stefan Schmidt]

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