Channel.c: Dropping incompatible voice frame on SIP/PW06622-00000681 of format gsm since our native format has changed to (u law)

channel.c: Dropping incompatible voice frame on SIP/PW06622-00000681 of format gsm since our native format has changed to (u
law)

Native format is the format that doesn’t require transcoding for the channel SIP/PW06622-00000681 in your case ulaw,

What would be the impact…I do calling through PRI within India only…will it put any issue on calling

Asterisk it is just dropping which are incompatible, due this in an excesive way could add extra load to the server, beside that I dont think any other issue

I think yes…because the load was on server around 900…shoking

But Sir how can i fix this because of this load was too high…

@asteriskpath You need to stop creating a thread over and over. The top 7 threads are all you. I’m going to delete some of them. Continue discussion in here if you want unless it is truly a completely different problem.

1 Like

I am soo sorry actually I am very new asterisk and fortunately/unfortunately I am managing infra of 25-30 servers…

yes you delete other thread and please help me here only…thanks a lot you everyone

@asteriskpath it seems you have a lot doubts and confusion about Asterisk, I suggest you hire an Asterisk consultant or be patient on the forum and wait the comunity help you, if you properly post the necesary information and logs abut your issue. But you cant flood the forum with your servers issue, first wait one or 2 or issue get answer and then continue posting

To add to that you also need to provide information - such as Asterisk version, what is going on, the specific problem you are experiencing.

sure sir I will take care of this for sure.

version 11 series is getting used

So basically the calls were not getting connected or were getting dropped after connecting. when I saw on cli I found the below alert message and the load was on server very very high approx 700+ was the load avereage. the version is beign used is asterisk 11.17

ERROR[29125][C-0000129c]: pbx.c:10463 ast_pbx_outgoing_exten: Unable to start PBX on SIP/TP03744-00001601
WARNING[9995]: tcptls.c:742 ast_tcptls_server_root: Unable to launch helper thread: Resource temporarily unavailable
WARNING[29126][C-0000129d]: pbx.c:6869 ast_pbx_start: Failed to create new channel thread
ERROR[29126][C-0000129d]: pbx.c:10463 ast_pbx_outgoing_exten: Unable to start PBX on SIP/TP03744-00001602

and just after some time on same server getting started below exception as well

channel.c: Dropping incompatible voice frame on SIP/PW06622-00000681 of format gsm since our native format has changed to (u
law)

Did you check what was using the CPU and causing the load average? As well Asterisk 11 is no longer supported, and had security releases after 11.17.0 to fix security issues.

yes the load average was too high that was approx 700-800 which usually remain 1.5-2.5 CPU was also very high 85-110%

we are having 5-6 servers of asterisk 11 only do you suggest to upgrade to newer version or can apply some security fix??

That didn’t answer my question - did you investigate what was causing the high load or are you assuming it is Asterisk? High load on a system will cause problems in Asterisk, but Asterisk itself may not be the cause.

One the threads I deleted also mentioned you believed you had been hacked - did you investigate and narrow that down?

You have to investigate further.

On load part: there was shell script running through which people pull out the recording files and that usually increase the load hence I stopped that script immediately and load came down…but again suddenly after couple of minutes load got increase and back to 700-800 probably the script i killed would be taking time.

yes i have bit doubt on hacking as well because I got a code in my extenison.conf file which is not written by me
[from-pstn]

exten =>6488751,1,Answer()
exten =>6488751,n,Set(CALLFILENAME=${CALLERIDNAME}-${CALLERID(name)}-${EXTEN}-${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)})
exten =>6488751,n,MixMonitor(${CALLFILENAME}.wav)
exten =>6488751,n,monitor(wav,${CALLFILENAME},m)
exten =>6488751,n,Goto(from-pstn,6488751,1)
exten =>6488751,n,Hungup()

I bielve this code was creation a loop of incoming calls because the pri of this did was plugged on this server…however this incoming call loop issues code was written on other server that was actually my backup server…when I moved my process on backup server huge number of calls came on server and server got hanged…I had to reboot and then I stop this code in exention.conf…
But I am not able to digest hacking because my server is not on public IP that is local network only …

by the way thanks for ur step by step support

this part of your dialplan it is the one who make the loop

Yes and I believe this will do anything with system…it can hang the system easly??

You server resources issues is something more than Asterisk issue, Asterisk it is only one of the many softwares that malfunction due to low system resources, hire a server admin who can find out what it is consuming your system resources