vicidial.org

by **albokos** » Tue Oct 20, 2009 8:53 am

Hi everybody my vicidial install is working fine. something unfortunatly is me. when my agents are speaking, clients on the other side of the line complains that the sound is bad as if we were calling them from far away. On the other side my agents have a perfect sound from their headsets I check my config and everything seems ok.

I am using g729 through my sip trunk as my provider request it.

my server is a P IV 2 Ghz; RAM (256); 40 Gb HDD, it is a test hardware.

I would like to have a better quality. could someone help me on this?????

by **brett05** » Tue Oct 20, 2009 9:30 am

what is your asterisk version ?
what is your vicidial version ?
what is your OS used ?
what is your sip config or what you have put in your admin-->trunk as config ?
do you see good your g729 in asterisk CLI by the use of show translation;other thing vérif when you call in your asterisk CLI to see if you use the g729 with your asterisk and your provider channel by the use of "sip show channel"
what you use sip or iax?
you need to use only transcoding g729 between your asterisk box and your provider then tranfer it to alaw and ulaw to your softphone so if you use a full version of eyebeam for exemple try to disable the codec g729 in his option and let only the ulaw and alaw
what codec do you have use the free one or the other with licence from diguim
what is your bandwith you use ?
do you use nat behind your phones ?
and finnaly try to post your CLI here
sorry they are many way to resolve your problem
:wink:

by **albokos** » Tue Oct 20, 2009 9:52 am

Thanks for reply

this is my config

os: debian 5.02
asterisk 1.4.26
vicidial svn trunk

I'm using sip trunk

Code: Select all: [siptrunk] type=friend username=myusername secret=mysecret host=sip.provider.example qualify=no canreinvite=no nat=yes //AND YES MY SERVER AND PHONES ARE BEHIND NAT dtmfmode=rfc2833 disallow=all allow=g729 insecure=port,invite

i am using the free version of g729 codec and it is loading properly in asterisk.

my bandwidth: 1Mbps (not guaranted)

And i don't get anything strange on my CLI but will post it as soon as i'm authorized to test it again.

So that's it hope i answered all your questions waiting your opinion

by **gardo** » Tue Oct 20, 2009 11:23 am

If you're using a P IV 2 Ghz; RAM (256), for your test hardware, I'd recommend you get a hardware timer card for Asterisk. This will greatly improve your audio quality since your test hardware is not powerful enough for the software based timer to do it's job properly.

Please post some of the output of "zttest -v".

by **albokos** » Tue Oct 20, 2009 11:54 am

The output of zttest -v

Code: Select all: 8192 zaptel samples in 8191.696 system clock sample intervals (99.996%) 8192 zaptel samples in 8191.224 system clock sample intervals (99.991%) 8192 zaptel samples in 8191.528 system clock sample intervals (99.994%) 8192 zaptel samples in 8191.536 system clock sample intervals (99.994%) 8192 zaptel samples in 8191.576 system clock sample intervals (99.995%) 8192 zaptel samples in 8191.528 system clock sample intervals (99.994%) 8192 zaptel samples in 8191.567 system clock sample intervals (99.995%) 8192 zaptel samples in 8191.592 system clock sample intervals (99.995%) 8192 zaptel samples in 8191.600 system clock sample intervals (99.995%) 8192 zaptel samples in 8191.584 system clock sample intervals (99.995%) 8192 zaptel samples in 8191.576 system clock sample intervals (99.995%)

i am not using a zaptel card do you think that if i increased the amount of memory i will get a better quality?, i'm afraid my boss will refuse to invest on a timer card.

by **brett05** » Tue Oct 20, 2009 12:03 pm

yes sure with card timing you will have a better voice
or try to change other g729 codec?

http://www.x100p.com/products/FXO.php

by **albokos** » Tue Oct 20, 2009 1:09 pm

thanks for the answer but since i get such a card, it will take a while can you give me some advices to enhance the performance???

Thank you again for your help.

by **ruben23** » Tue Oct 20, 2009 1:33 pm

Do answer the Info's asked by Brett05, it would help make your setup more clear to us to assist you... :wink:

by **albokos** » Wed Oct 21, 2009 6:13 am

sorry for the late answer but i just have an hour to test my system every day so i have to wait til my time come and i will post my CLI output. Thanks for your help i will post the info you request in a moment

by **albokos** » Wed Oct 21, 2009 10:11 am

Here is my CLI at loggin:

Code: Select all: [Oct 21 15:00:37] -- Remote UNIX connection [Oct 21 15:01:02] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:01:02] Found [Oct 21 15:01:02] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:01:02] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:01:02] Found [Oct 21 15:01:02] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:01:02] ERROR[23514]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer [Oct 21 15:01:02] ERROR[23514]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Oct 21 15:01:02] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 21 15:01:07] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:01:07] Found [Oct 21 15:01:07] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:01:07] ERROR[23522]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer [Oct 21 15:01:07] ERROR[23522]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Oct 21 15:01:07] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 21 15:01:07] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 21 15:01:09] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:01:09] Found [Oct 21 15:01:09] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:01:14] > Channel SIP/cc100-09d3b360 was answered. [Oct 21 15:01:14] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 21 15:01:14] -- Executing [8600051@default:1] MeetMe("SIP/cc100-09d3b360", "8600051|F") in new stack [Oct 21 15:01:14] == Parsing '/etc/asterisk/meetme.conf': [Oct 21 15:01:14] Found [Oct 21 15:01:14] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Oct 21 15:01:14] Found [Oct 21 15:01:14] -- Created MeetMe conference 1023 for conference '8600051' [Oct 21 15:01:14] -- <SIP/cc100-09d3b360> Playing 'conf-onlyperson' (language 'fr') [Oct 21 15:01:15] NOTICE[23536]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 received from '172.16.1.140' [Oct 21 15:01:15] NOTICE[23536]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 received from '172.16.1.140' [Oct 21 15:01:15] NOTICE[23536]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 received from '172.16.1.140'

and when in call:

Code: Select all: 15:09:53] -- Called siptrunk/33490185883 [Oct 21 15:09:53] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:09:53] Found [Oct 21 15:09:53] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:09:53] -- Executing [33490185897@default:1] AGI("Local/33490185897@default-873f,2", "agi://127.0.0.1:4577/call_log") in new stack [Oct 21 15:09:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Oct 21 15:09:53] -- Executing [33490185897@default:2] Dial("Local/33490185897@default-873f,2", "SIP/siptrunk/33490185897||To") in new stack [Oct 21 15:09:53] -- Called siptrunk/33490185897 [Oct 21 15:09:55] -- SIP/siptrunk-09dd2b38 is making progress passing it to Local/33490185883@default-0e15,2 [Oct 21 15:09:55] -- SIP/siptrunk-09ef6680 is making progress passing it to Local/33490185897@default-873f,2 [Oct 21 15:09:58] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:09:58] Found [Oct 21 15:09:58] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:09:58] -- Executing [33490185900@default:1] AGI("Local/33490185900@default-e0b5,2", "agi://127.0.0.1:4577/call_log") in new stack [Oct 21 15:09:58] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Oct 21 15:09:58] -- Executing [33490185900@default:2] Dial("Local/33490185900@default-e0b5,2", "SIP/siptrunk/33490185900||To") in new stack [Oct 21 15:09:58] -- Called siptrunk/33490185900 [Oct 21 15:09:59] -- SIP/siptrunk-09dd2b38 answered Local/33490185883@default-0e15,2 [Oct 21 15:09:59] > Channel Local/33490185883@default-0e15,1 was answered. [Oct 21 15:09:59] -- Executing [8368@default:1] Playback("Local/33490185883@default-0e15,1", "sip-silence") in new stack [Oct 21 15:09:59] -- <Local/33490185883@default-0e15,1> Playing 'sip-silence' (language 'en') [Oct 21 15:09:59] WARNING[24468]: file.c:1273 waitstream_core: Unexpected control subclass '-1' [Oct 21 15:09:59] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 21 15:09:59] -- Executing [8368@default:2] AGI("Local/33490185883@default-0e15,1", "agi://127.0.0.1:4577/call_log") in new stack [Oct 21 15:09:59] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Oct 21 15:09:59] -- Executing [8368@default:3] AGI("Local/33490185883@default-0e15,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack [Oct 21 15:09:59] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Oct 21 15:09:59] -- SIP/siptrunk-09d90f30 is making progress passing it to Local/33490185900@default-e0b5,2 [Oct 21 15:09:59] -- Executing [h@default:1] DeadAGI("Local/33490185883@default-0e15,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0") in new stack [Oct 21 15:09:59] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0 completed, returning 0 [Oct 21 15:09:59] == Spawn extension (default, 33490185883, 2) exited non-zero on 'Local/33490185883@default-0e15,2' [Oct 21 15:10:00] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Oct 21 15:10:00] -- Executing [8368@default:4] AGI("SIP/siptrunk-09dd2b38", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack [Oct 21 15:10:00] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Oct 21 15:10:00] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Oct 21 15:10:00] -- Executing [172*016*001*120*8600051@default:1] Goto("SIP/siptrunk-09dd2b38", "default|8600051|1") in new stack [Oct 21 15:10:00] -- Goto (default,8600051,1) [Oct 21 15:10:00] -- Executing [8600051@default:1] MeetMe("SIP/siptrunk-09dd2b38", "8600051|F") in new stack [Oct 21 15:10:02] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:10:02] Found [Oct 21 15:10:02] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:10:02] == Parsing '/etc/asterisk/manager.conf': [Oct 21 15:10:02] Found [Oct 21 15:10:02] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 21 15:10:02] ERROR[24489]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer [Oct 21 15:10:02] ERROR[24489]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Oct 21 15:10:02] == Manager 'sendcron' logged off from 127.0.0.1 asterisk*CLI>

What do you think of this?? hope it will help !!!

by **brett05** » Thu Oct 22, 2009 12:49 am

first have you patch your asterisk
2=have you use transfer codec from asterisk to xlite as this:
g729-->ulaw or alaw --->"softphone(xlite)"
|
|
|_______________________"asterisk server"

so in your xlite juste enable the good codec and not g729 if you use a full version of eyebeam.
note:if your provider just allaw g729 and if for exemple we hope have a good voice quality and we have not a asterisk server ,so here we will try a full version of softphone with codec G729 as eyebeam for exemple,but in our exemple here we have a asterisk server that will receive a sip voip from our provider with codec g729 so here we need just transcoding this sip voip with add of g729 in our server so here our server will understand that we have a codec too.

Provider------->Asterisk"G729"-------->Xlite"ULAW OR ALAW (G711) "
|
|
|___________ G729

after for this error:
rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 received from '172.16.1.140'
and it come from the Using dtmfmode=rfc2833 whith codec G729A.
or This is because those 2 devices for some reason decide to send DTMF.
finnaly for the other error i have not a idea it come from asterisk and maybe you have not compliled good your server .

by **albokos** » Thu Oct 22, 2009 6:33 am

Thank you for your reply if i understand well, what i have to do is to disable g729 from xlite and replace it with alaw ou ulaw, activate it only on asterisk so then the server could transcode the signal. and then suppress the option

Code: Select all: dtmfmode=rfc2833

from the dialplan.

Am i on the way??????

for this error

Code: Select all: ERROR[24489]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

i think it is due to a synch problem with the database (a bug i think). don't think it is a serious issue (but maybe i'm wrong).

Could you give me some more info about patching asterisk?? what patch do i need ?? and what bugs does it fix.

thank you.

by **albokos** » Sat Oct 24, 2009 8:29 am

Anybody????

by **brett05** » Sat Oct 24, 2009 9:36 pm

strange ??
have you try to install other frech asterisk with apache and vicidial in debian or ubuntu ?
have you compile good your kernel ?
have you compile good your cpan ?
what libpri and zaptel version you have setup ?
try to use the setup found in ubuntu manual for vicidial with asterisk 1.4.
and finnaly show your step for installing asterisk

by **albokos** » Sun Oct 25, 2009 7:45 pm

i've reinstall asterisk and vicidial on a PIV, 3Ghz with hyperthreading, 512 RAM, and 40Gb HDD. my os is debian again

Compile the kernel??????

my cpan is well compiled i think! how can i check it???

i'm using zaptel 1.4.12 and libpri 1.4 too

to install asterisk and vicidial, i followed the steps describe for ubuntu and to install asterisk i did:

Code: Select all: tar -zxvf asterisk-1.4.26...... cd asterisk..... make clean ./configure make menuselect make make install make samples make config

to check if it is a system failure, i called a client directly from eyebeam with the account i use for sip trunking and gess what!!! the client complained about the bad sound. so i think the problem comes from the network.

i set the tos_audio parameter of asterisk to ef, hope it will help. unfortunatly i can not check my cisco router config because it is managed by an external operator. i'm a bit desperate i'm so close !!!!!!

by **gmcust3** » Fri Nov 13, 2009 11:38 am

Did u try Sangoma VoiceTime USB stick ?

vicidial.org

Bad sound quality

Bad sound quality

thanks

thanks

thanks

thanks

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