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
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