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Hold Music during call

PostPosted: Sat Nov 21, 2009 6:21 am
by phil_discount
Hello,

I've got a strange problem.
sometimes when an agent has got a live call, the agent gets holdingmusic and the customer hears a voice text, don't know what really.

I searched message log and some other logs,, but i found nothing special.

we are using PRI to dial out and the agents are connected to our tksystem with a PRI too.

Anybody an idea what this can be?
or where i can search for the problem?

regards
philip

PostPosted: Sat Nov 21, 2009 6:41 am
by Michael_N
Is this problem specific for a campaign?

It seems like a survey.

PostPosted: Sat Nov 21, 2009 11:12 am
by phil_discount
no survey, on each campaign

PostPosted: Sun Nov 22, 2009 3:18 am
by williamconley
you'd have to find an astguiclient log and/or cli to explain this behavior. too many possibilities. have to catch it happening.

PostPosted: Mon Nov 23, 2009 8:37 am
by phil_discount
OK, today i observed the agents and looked at asterisk CLI.

when the music come during a call, the CLI has no output..
for example is playing sip-silence or else...
the call was 13 minutes long and the agent and customer hear about 4x times the music (for about 3-6 seconds) after music, the agent and customer can talk on....very strange!! :-)
in action_full.date and listen.date and action_send.date LOGs aren't any entries during call.

i don't know if is a problem but asterisk 1.2.30.4 is installed and in admin/server section i read 1.2.26.2. now i change it, but i have to reboot the system in the evening.

perhaps the music (can't describe it, really slow music (-: ) will be played from the PRI carrier?

regards
philip

PostPosted: Mon Nov 23, 2009 9:16 am
by mflorell
I've never heard of ViciDial doing anything like this.

How often does this occur?

the asterisk version would treat 1.2.30.4 and 1.2.26.2 exactly the same, so there is really no reason to change that.

PostPosted: Mon Nov 23, 2009 9:52 am
by phil_discount
in a long call sometimes 4x times.
it occurs 50-100x times the whole day for 30 agents..

not campaign specific...tomorrow i will only call over SIP and check if the problem still occur.

PostPosted: Tue Nov 24, 2009 9:55 am
by phil_discount
unbelievable, the problem occurs only with one provider over PRI.
i will create a service ticket ... if i get feedback, i will report it

PostPosted: Sat Nov 28, 2009 9:18 am
by phil_discount
my provider has no errors in debugmodus.
now i know which music will be played during call
"macroform-cold_day.gsm"
does anybody know where the music come from?

regards
philip

PostPosted: Sat Nov 28, 2009 1:42 pm
by mflorell
Not from ViciDial :)

PostPosted: Sun Nov 29, 2009 5:45 am
by phil_discount
the file is located in asterisk /usr/src/asterisk-1.2.30.4/sounds/moh/macroform-cold_day.gsm ..

but i don't know why it is played :-/

EDIT:
should i rename it? so perhaps i get an error in CLI

PostPosted: Mon Nov 30, 2009 5:11 pm
by phil_discount
it's not the cold_day music...
i think the music will be played by the provider...

the question is why?
and why the music is only played sometimes and during a call...

i don't understand

PostPosted: Tue Dec 01, 2009 9:22 am
by phil_discount
i found out something new.

the problem still occurs on all providers, so it can't be the provider.
i tried dialing over SIP and PRI.

the holdmusic vom asterisk is it
/var/lib/asterisk/mohmp3/reno_project-system.wav

perhaps it's a connection problem between tksystem and asterisk.
tomorrow i will try another port on the sangoma PRI card...

in the dialplan i use 35,Tto,
can i remove Tt in dialplan or is it used by vicidial?
Code: Select all
t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf)
If you have set the variable GOTO_ON_TRANSFER then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_TRANSFER=woohoo^s^1); works with both t and T
WARNING: GOTO_ON_TRANSFER does not exist in any version of ASTERISK and will not! the variable is called GOTO_ON_BLINDXFR see http://svn.digium.com/view/asterisk?rev=5495&view=rev and http://bugs.digium.com/view.php?id=4056 for details. THX to the person who shared the information above!
T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf)


in CLI i found something
Code: Select all
root@ka-vici-dialer:/var/lib/asterisk/mohmp3# tail -n 300000 /var/log/astguiclient/screenlog.0 | grep 3968
    -- Executing MeetMe("Local/8600055@default-3968,2", "8600055|F") in new stack
       > Channel Local/8600055@default-3968,1 was answered.
    -- Executing AGI("Local/8600055@default-3968,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- Executing Dial("Local/8600055@default-3968,1", "SIP/colt/025626012|35|tTo") in new stack
    -- SIP/colt-082bf590 is making progress passing it to Local/8600055@default-3968,1
    -- SIP/colt-082bf590 is making progress passing it to Local/8600055@default-3968,1
    -- SIP/colt-082bf590 is making progress passing it to Local/8600055@default-3968,1
    -- SIP/colt-082bf590 answered Local/8600055@default-3968,1
    -- Started music on hold, class 'default', on Local/8600055@default-3968,1
    -- Stopped music on hold on Local/8600055@default-3968,1
    -- Started music on hold, class 'default', on Local/8600055@default-3968,1
    -- Stopped music on hold on Local/8600055@default-3968,1

PostPosted: Wed Dec 02, 2009 1:19 am
by mflorell
the reason you want a t or T in the Dial string is to force an Asterisk-bridged call, if you don't put one of those in Asterisk may try to native bridge the call and you will loose any ability to control or record the call.

PostPosted: Thu Dec 03, 2009 5:41 am
by phil_discount
ok i think i found the problem.
since two days the problem is fixed.

i changed the following:

SIP.conf:
canreinvite=no

zapata.conf
park call no
callreturn no
3waycall no

zaptel.conf (changed the second value from 0 to 1 on all spans)
span=1,1,0,ccs,hdb3,crc4

i would like to know which of the changes has fixed the problem.
so i have to change one value and wait :-)

can anybody tell me how i can restart asterisk/wanpipe/zaptel correctly?
because everytime i change something, i reboot the whole system :-)
i'm using vicibox standard installation.

asterisk -r -> stop now
wanrouter stop
how i stop zaptel?

wanrouter start
how i start zaptel?
/usr/sbin/asterisk (which parameters)

regards
philip

PostPosted: Thu Dec 03, 2009 10:38 am
by mflorell
If zaptel is really involved then the only sure way of reloading everything is to reboot.

If it's just asterisk then you can "stop now" and use the start_asterisk_boot.pl script to restart Asterisk properly.

PostPosted: Thu Dec 03, 2009 1:08 pm
by phil_discount
thanks, next week i will change every value - day by day - when i found out, i will report.

thanks to all

PostPosted: Sat Jun 25, 2011 8:00 am
by phil_discount
problem was in features.conf

blindxfer => CC
;disconnect => B

no more holdmusic with this settings

PostPosted: Sat Jun 25, 2011 9:08 am
by mflorell
Thanks for posting your solution!

Re: Hold Music during call

PostPosted: Thu Jun 02, 2016 3:12 am
by praveenappadurai
Hello,

I Had a same Problem. Agents are in calls sometimes they hear the Hold music.

I saw here Phil posted a solution it worked for him.He posted like this

blindxfer => CC
;disconnect => B

In my settings it looks like below

blindxfer => CC
disconnect => B

So can i add the Comment and test. Will it affect other Process or activities?

Please Post me a Solution.

Re: Hold Music during call

PostPosted: Wed Jun 22, 2016 4:36 pm
by williamconley
1) Welcome to the Party! 8-)

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Probably won't alter or interfere with anything else. But: That's why you'd be testing. Since your system appears to be "odd", it could be odd in other ways as well. But since we know nothing of your system ... I'll hold on to the opinion that your system dates to the moment when this bug may have existed. So fix it and pay attention in case you have to put it back.