provided by Star2Billing S.L.

How can I stop these errors its create huge files every day but does not seem to slow my dual quad core clovertown processors.

Sep 4 17:54:03 NOTICE[15372] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible.
Client IP: 91.13.102.245
Sep 4 17:54:03 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:03 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:03 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:04 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:04 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:04 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:04 NOTICE[15265] frame.c: Dropping extra frame of G.729 since we already have a VAD frame at the end
Sep 4 17:54:08 WARNING[15436] acl.c: Cannot create socket
Sep 4 17:54:08 WARNING[15436] channel.c: Channel allocation failed: Can't create alert pipe!
Sep 4 17:54:08 WARNING[15436] chan_sip.c: Unable to allocate SIP channel structure
Sep 4 17:54:08 NOTICE[15436] app_dial.c: Unable to create channel of type 'SIP' (cause 0 - Unknown)
Sep 4 17:54:08 WARNING[10642] rtp.c: Unable to allocate socket: Too many open files
Sep 4 17:54:08 WARNING[10642] channel.c: Channel allocation failed: Can't create alert pipe!
Sep 4 17:54:08 WARNING[10642] chan_sip.c: Unable to allocate SIP channel structure
Sep 4 17:54:08 NOTICE[10642] chan_sip.c: Unable to create/find channel
Sep 4 17:54:09 WARNING[15444] res_agi.c: Unable to create toast pipe: Too many open files
Sep 4 17:54:09 ERROR[15444] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files
Sep 4 17:54:09 WARNING[15444] cdr_csv.c: Unable to write CSV record to account file '901381772155274' : Too many open files
Sep 4 17:54:09 ERROR[15444] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open fil
es
Sep 4 17:54:10 WARNING[10642] channel.c: Channel allocation failed: Can't create alert pipe!
Sep 4 17:54:10 WARNING[10642] chan_sip.c: Unable to allocate SIP channel structure
Sep 4 17:54:10 NOTICE[10642] chan_sip.c: Unable to create/find channel
Sep 4 17:54:10 WARNING[15449] acl.c: Cannot create socket
Sep 4 17:54:10 WARNING[15449] channel.c: Channel allocation failed: Can't create alert pipe!
Sep 4 17:54:10 WARNING[15449] chan_sip.c: Unable to allocate SIP channel structure
Sep 4 17:54:10 NOTICE[15449] app_dial.c: Unable to create channel of type 'SIP' (cause 0 - Unknown)
Sep 4 17:54:10 WARNING[15458] res_agi.c: Unable to create toast pipe: Too many open files

"messages" 5277321L, 600147629C

THIS IS NOT ASTERISK SUPPORT FORUM!!!

The first errors (dropping extra frame) could be A2B related in that a prefix needs to be set in the trunk to tell the sip provider to turn off VAD support. eg; 89981XXXXXXXXXX

if the provider does not allow this prefix then you have to shop around because this basically means your customers have audio issues.

You are right about the second set of errors seems to be an asterisk issue resulting from the VAD issue.

Since I'm at it, I'll suggest you increase your file descriptors by placing at the end of /etc/security/limit.conf (f they do not exist or change the numbers to match).
root soft nofile 4096
root hard nofile 8196
asterisk soft nofile 4096
asterisk hard nofile 8196

i must repeat:
THIS IS NOT ASTERISK SUPPORT FORUM!!!!
in your answer i can't see any tips for Asterisk2Billing, i can see only asterisk here... i remind again - VAD is NOT 'A2B related', never was and never will!!!! Go to Asterisk forum(s) with this questions and please respect community... Thank you.

Asterisk and A2B are entwined and if Digium refuses to fix the issues that affect our business then we have to do it somewhere else.

I've just had enough of these "dropping frames" and did a research and found a fix in the digium bug tracker but it didn't work very well. To make it complete, I had to modify it a bit. Now there is no issue with VAD. (asterisk 1.4)

In frame.c change:

if (s->flags & AST_SMOOTHER_FLAG_G729) {
if (s->len % 10) {
ast_log(LOG_NOTICE, "Dropping extra frame of G.729 since we already have a VAD frame at the end\n");
return 0;
}
}

To:

if (s->flags & AST_SMOOTHER_FLAG_G729) {
if (s->len < 10) {
return 0;
}
if (s->len % 10) {
s->len -= (s->len % 10);
}
}

Hi Vulcan,

I tried your VAD fix for Asterisk 1.4.21.2 and it does not work in our setup. Which version did you run it on successfully?

Thanks
Raja

It was used on 1.4x but this patch was abandoned and the switch was made back to ulaw and ilbc which was working quite well and still does. G729 isn't all that compared to the voice quality your customers get with ulaw. As far as bandwith issues I haven't noticed any. You just need a good fast connection and memory.

I'm taking cue from two big retail companies, Callcentric and Broadvoice, all they use is ulaw. Not to mention fax can be transmitted on ulaw.

I am no asterisk guru, but I killed myself today trying to fix this same error. And incredibly it went away when I set canreinvite=yes on the [general] context of sip.conf

Actually, it is not incredible, there is a good reason for it, but that I leave for the interested to research on.

KingDavid

That is good for wholesale, but running calling card, asterisk needs to stay in the RTP. When A2B gives out warnings prompts about balance and time left customers will not hear it. Again, Asterisk needs to stay in RTP to disconnect call if no rtp activity for a designated periond of time.


It's a possible billing issue with canreinvite=yes.