Here is the bug report https://issues.asterisk.org/view.php?id=19171
Please add a comment to the bug indicating that you are also experiencing the issue with asterisk 1.4.35 to 1.4.41 -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Satish Patel Sent: Wednesday, May 04, 2011 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.4.35 to 1.4.41 Look like codec mismatch issue. -- Sent from my iPhone On May 3, 2011, at 9:55 PM, Jerry Geis <[email protected]> wrote: > Under 1.4.35 I get this message printed MANY times > [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame > type 4, while native formats is 0x1000 (g722)(4096) read/write = > 0x1000 (g722)(4096)/0x1000 (g722)(4096) > [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame > type 4, while native formats is 0x1000 (g722)(4096) read/write = > 0x1000 (g722)(4096)/0x1000 (g722)(4096) > [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame > type 4, while native formats is 0x1000 (g722)(4096) read/write = > 0x1000 (g722)(4096)/0x1000 (g722)(4096) > [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame > type 4, while native formats is 0x1000 (g722)(4096) read/write = > 0x1000 (g722)(4096)/0x1000 (g722)(4096) > [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame > type 4, while native formats is 0x1000 (g722)(4096) read/write = > 0x1000 (g722)(4096)/0x1000 (g722)(4096) > > Under 1.4.41 I get an error and hang up doing the exact same thing. > > All I am doing Is calling a cell phone over the PRI then dialing my > SIP/524 extension. > > > This is from 1.4.35 > > Channel DAHDI/18-1 was answered. > -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf > ("DAHDI/18-1", "1?smvoice_callprogress|3:smvoice_callprogress|2") in > new stack > -- Goto (smvoice-dialout,smvoice_callprogress,3) > -- Executing [smvoice_callprogress@smvoice-dialout:3] AGI("DAHDI/ > 18-1", "smvoice) in new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice > -- Playing '/home/silentm/record/please_press/ > one_to_call.' (escape_digits=0123456789*#) (sample_offset 0) > [May 3 21:47:38] DTMF[21746]: channel.c:2368 __ast_read: DTMF end > '1' received on DAHDI/18-1, duration 0 ms > [May 3 21:47:38] DTMF[21746]: channel.c:2423 __ast_read: DTMF end > accepted without begin '1' on DAHDI/18-1 > [May 3 21:47:38] DTMF[21746]: channel.c:2434 __ast_read: DTMF end > passthrough '1' on DAHDI/18-1 > -- Playing '/tmp/smvoice.21747_0' (escape_digits=0123456789#) > (sample_offset 0) > [May 3 21:47:41] ERROR[21746]: utils.c:968 ast_carefulwrite: write > () returned error: Broken pipe > -- AGI Script smvoice completed, returning 0 > -- Executing [smvoice_dial_goto_voicemail@smvoice-dialout:1] Dial > ("DAHDI/18-1", "SIP/524|30|tT") in new stack > -- Called 524 > [May 3 21:47:41] WARNING[21746]: channel.c:3782 > ast_channel_make_compatible: No path to translate from SIP/ > 524-00000001(4096) to DAHDI/18-1(4) > [May 3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked > to transmit frame type 4, while native formats is 0x1000 (g722) > (4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) > [May 3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked > to transmit frame type 4, while native formats is 0x1000 (g722) > (4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) > > Is this a problem with 1.4.41 or my Polycom HD Voice phone with g722 > codec or both? > (again - it works under 1.4.35 just prints a message many many times) > > Jerry > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
