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Summary:ASTERISK-14384: Early media causes "Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8)......"
Reporter:Alec Davis (alecdavis)Labels:
Date Opened:2009-06-26 07:19:47Date Closed:2010-01-06 14:11:17.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
duplicatesASTERISK-18063 Flooding with [Jun 24 19:33:17] WARNING[6995]: chan_sip.c:6213 sip_write: Asked to transmit frame type g726, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
is related toASTERISK-18325 "Asked to transmit frame type" slows down all the calls
Environment:Attachments:
Description:Calling from a SIP phone to a local device or over an IAX trunk that sends Early Media causes many messages on screen, and no audio.

Console Screen fills with, and no audio is heard.

[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)

Calling from FXS port to FXS port works great, a triple ring is heard at callers handset.

This used to work when I originally submitted https://issues.asterisk.org/view.php?id=14504

Another recent bug that may be related:
https://issues.asterisk.org/view.php?id=14310


****** ADDITIONAL INFORMATION ******

Another bug/feature that easily reproduces this is https://issues.asterisk.org/view.php?id=14504

dialplan code:
exten => 89,1,Dial(DAHDI/4,20,r(vodaring))
exten => 89,n,Voicemail(89,u)

/etc/indications.conf, add line below in your loadzone.
vodaring = 400+450/400,0/200,400+450/200,0/200,400+450/400,0/2000

Asterisk SVN-trunk-r203569M
Comments:By: Alec Davis (alecdavis) 2010-01-06 02:42:16.000-0600

Trunk fixed in ASTERISK-13606

Branches may still have the problem, I believe it's due to the early
ast_channel_make_compatible when asterisk is generating musicback or ringback.

app_dial.c:wait_for_answer
should look like this<pre>
...
           /* Turn off hold music, etc */
           if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK)) {
                   ast_deactivate_generator(in);
                   /* If we are calling a single channel, and not providing ringback or music, */
                   /* then, make them compatible for in-band tone purpose */
                   <b><u>ast_channel_make_compatible(outgoing->chan, in);</u></b>
           }</pre>


By: Alec Davis (alecdavis) 2010-01-06 03:48:57.000-0600

related: ASTERISK-14999 No streaming musiconhold when using dial command

Using trunk SVN-trunk-r237920M, and undoing the early ast_channel_make_compatible() fix from ASTERISK-13606, I can confirm the problem and get the same.
[Jan  6 22:40:20] WARNING[15914]: chan_sip.c:6526 sip_write: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)
[Jan  6 22:40:21] WARNING[15914]: chan_sip.c:6526 sip_write: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)

What I forgot to add in the previous note ~116103, was when the called party answers, ast_channel_make_compatible() is executed again.
By: Alec Davis (alecdavis) 2010-01-06 04:26:31.000-0600

Became a regression with asterisk commit 186525

I take back what I suggested about branches, they are not affected.

This can now be closed.
By: Leif Madsen (lmadsen) 2010-01-06 14:11:16.000-0600

Closed per the reporter on IRC.