[asterisk-users] asterisk 1.4.35 to 1.4.41

[Eric Wieling]

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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 <geisj at pagestation.com> 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 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
>
> --
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