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Summary:ASTERISK-18674: CLONE - Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
Reporter:Nicholas Barnes (nab)Labels:
Date Opened:2011-10-05 09:32:55Date Closed:2011-11-01 10:23:23
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:1.8.3 Frequency of
Occurrence
Related
Issues:
is a clone ofASTERISK-17502 Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
Environment:Attachments:
Description:Astersik flooding with message :
chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)

When user test dial at outtrunk:
Dial(SIP/somenumberhere@outtrunk,40,m);
i got many warninig messages like above. When we remove m(music on hold) parameter everything works fine:
Dial(SIP/somenumberhere@outtrunk,40);

We have installed moh files in alaw and g729 formats, and we using g729 codec from: <link removed by lmadsen>





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

part of sip.conf:

[test]
type=friend
callerid="test"
username=test
host=dynamic
secret=test
dtmfmode=info
canreinvite=yes
nat=yes
qualify=yes
context=outgoing
disallow=all
allow=alaw

[outtrunk]
type=peer
host=21.21.12.45
canreinvite=no
nat=yes
sendrpid=yes
disallow=all
allow=g729
Comments:By: Nicholas Barnes (nab) 2011-10-05 09:34:24.372-0500

This was previously closed, however I can reliably replicate the problem and it's causing me issues in the real world!!


I can replicate this problem as follows:

A call arrives at our box in alaw format [1]
We answer the call [2]
Play a message (Playback(custom/welcome)) [3]
Dial a handset which only supports/permits GSM [4]

The error below is repeated for every frame of MoH until the handset is answered or the call is terminated.

[Oct 5 15:22:13] WARNING[16456]: chan_sip.c:6341 sip_write: Asked to transmit frame type alaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)

Once the call is answered, everything works perfectly.

I also have an issue that Asterisk dumps core when this happens a lot.

[1] We have no control over this - it's alaw or nothing!
[2] With 'Answer(500)'
[3] The file 'custom/welcome.alaw' is played
[4] With 'Dial(SIP/device,,tm)' so the caller hears MoH.

By: Gregory Hinton Nietsky (irroot) 2011-10-07 03:47:22.029-0500

The fix is for 1.8.8 and above please try with the lattest version currently 1.8.8.0-rc1
By: Marc Purdon (bromont) 2011-10-12 21:34:14.931-0500

I can confirm the issue is resolved in 1.8.8

By: Gregory Hinton Nietsky (irroot) 2011-10-13 03:56:14.560-0500

Awesome thx for the feedback