[asterisk-users] asterisk 1.4.35 to 1.4.41

[Jerry Geis]

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 at 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 at 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 at 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