| Summary: | ASTERISK-17502: Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x | ||||||||||||
| Reporter: | Arkadiusz Malka (yarns) | Labels: | |||||||||||
| Date Opened: | 2011-03-03 05:55:56.000-0600 | Date Closed: | 2011-04-19 10:04:19 | ||||||||||
| Priority: | Minor | Regression? | No | ||||||||||
| Status: | Closed/Complete | Components: | Channels/chan_sip/CodecHandling | ||||||||||
| Versions: | 1.8.3 | Frequency of Occurrence | |||||||||||
| Related Issues: |
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| 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: Leif Madsen (lmadsen) 2011-03-03 14:06:45.000-0600 We can't support that third party code. Please take this issue up with them. By: Arkadiusz Malka (yarns) 2011-03-04 03:38:11.000-0600 We bought one channel license for g729 from digium. Asterisk behave the same way like with open source g729 codec. If you need order information to proof contact me. By: Arkadiusz Malka (yarns) 2011-03-04 03:40:06.000-0600 Can i get my money back :) By: Leif Madsen (lmadsen) 2011-03-08 11:57:11.000-0600 This is not the appropriate forum to as for refunds of commercial software. Please contact Digium directly. By: Leif Madsen (lmadsen) 2011-03-08 12:44:52.000-0600 Looks like because you're using MoH like that the system is probably requiring more licenses. Because you've purchased a commercial license you should take this up with Digium support directly. By: Arkadiusz Malka (yarns) 2011-03-26 00:52:16 I configured the trunk to use alaw, and my extension to use ulaw and found the same issue. This appears to be a bug in Asterisk, as it occurs even when G.729 is not used at all: [Mar 25 22:40:47] WARNING[22217]: chan_sip.c:6047 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) By: Leif Madsen (lmadsen) 2011-04-05 14:29:52 <pabelanger> leifmadsen: I usually see that when I don't have all the required codec modules loaded <pabelanger> leifmadsen: what if he tried directmedia=no By: Leif Madsen (lmadsen) 2011-04-19 10:04:19 No response from reporter. Presumed configuration error. By: Nicholas Barnes (nab) 2011-10-05 09:32:20.397-0500 I can replicate this problem: 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 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. | ||||||||||||