Sounds to me like inband vs rfc2833 issues. I found that one has to use the same codec throughout in order to make DTMF function and then use inband. This in turn forces you down the road of alaw or ulaw codecs.
On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings I get when I send DTMF from > Asterisk to the 2600: > > == Forcing Marker bit, because SSRC has changed > Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to > find a codec translation path from ilbc to ulaw > Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to > find a codec translation path from ilbc to ulaw > Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to > transmit frame type 1024, while native formats is 4 (read/write = 4/4) > Jun 26 17:53:52 WARNING[14248]: channel.c:2693 > ast_channel_make_compatible: No path to translate from > SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) > Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: > Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible > Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: > Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa > == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on > 'SIP/53061-92e0' > > The call drops. > > If I enable ILBC codec with Asterisk, here is what I get: > > == Forcing Marker bit, because SSRC has changed > Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: > Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP > (160)? > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > > The call continues with this error until I hang up. > > I have been adjusting the dial-peer dtmf settings in the 2600 and have > tried all the dtmf settings in Asterisk. > > Any guidance will be appreciated. > > Thanks. > > JR _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
