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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/381e8aa2/attachment.htm
Hi!> 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.I *think* there is/was an entry in the bug tracker on this. You might want to search https://issues.asterisk.org (also look for RTP issues with SSRC) and in the meanwhile you could reveal which version of Asterisk you are using. :) Philipp
This is the exit of "core show version": Asterisk 1.6.0.28 built by root @ AST on a i686 running Linux on 2010-06-28 12:21:24 UTC Obg, Rodrigo Lang. 2010/7/20 Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de>> Hi! > > > 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. > > I *think* there is/was an entry in the bug tracker on this. You might > want to search https://issues.asterisk.org (also look for RTP issues with > SSRC) and in the meanwhile you could reveal which version of Asterisk you > are using. :) > > Philipp > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/fe9c2218/attachment.htm
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