vicidial.org

Outbound calls get voice dalay. Landline to agent is almost ok, but agent to landline gets circa 0,5 sec dalay. Sometimes it gets worse - both directions get like 4-5 seconds from speaking to hearing. Lmao - i can call (myself) and have enough time to answer (myself) in a witty way

I gather:

1. Turn off recording.
2. Use g711
3. IAX
4. Use sata drive.
5. System load before and during calls.
6. Traceroute to my voip provider.
7. Hardware timing device.


Also i go using:
disallow=all
allow=all

allow=g729 and g711 doesn't seem to work - which doesn't make sense. If voip provider accepts only 729 and 711 than how come i can talk with allow all and not 729/711?

How do i check what codecs are installed and how to install them if there are none?

Code: Select all

show translation

anything with "-" you do not have (also anything not listed, you do not have)

it is worth noting that if you have not installed g729, you do not HAVE g729. it's not free, it's not open source, it will not "auto-install" with GoAutoDial or Vicibox Redux (nor did it with VicidialNOW or Vicibox classic).

Ok, issue resolved, hopefully permanently.

1. I have agent using g729 softphone.
2. Server has installed g729.
3. Voip provider accepts g729.


That means no transcoding between 1>2>3. Thats good, right (just pass the data on, no time wasted for "translation")?


I gather that there are 2 kinds of codecs:

codec_g729-ast14-gcc4-glibc-pentium4.so
codec_g729-ast14-icc-glibc-pentium4.so

where ast14 is asterisk version (cli says "Connected to Asterisk 1.4.27.1-1 RPM by demian@goautodial.com")

and gcc4/icc are, whatever they are

I'm gonna try both since i get (in the landline side) a lot of (and loud) static and also the voice (landline too) is very quiet.

Bump with new info.





So ok, I was too happy to check. Seems to me i go using ulaw all the time.

How do i get vici to run by g729.

Allow=g729 gives:


[May 15 12:40:42] == Parsing '/etc/asterisk/manager.conf': [May 15 12:40:42] Found
[May 15 12:40:42] == Manager 'sendcron' logged on from 127.0.0.1
[May 15 12:40:42] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-581f,2", "8600051|F") in new stack
[May 15 12:40:42] > Channel Local/8600051@default-581f,1 was answered.
[May 15 12:40:42] -- Executing [910817479546@default:1] AGI("Local/8600051@default-581f,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 15 12:40:42] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 15 12:40:42] -- Executing [910817479546@default:2] Dial("Local/8600051@default-581f,1", "SIP/0817479546@goautodial||tTo") in new stack
[May 15 12:40:42] -- Called 0817479546@goautodial
[May 15 12:40:42] -- SIP/goautodial-0000000c is circuit-busy
[May 15 12:40:42] == Everyone is busy/congested at this time (1:0/1/0)
[May 15 12:40:42] -- Executing [910817479546@default:3] Hangup("Local/8600051@default-581f,1", "") in new stack
[May 15 12:40:42] == Spawn extension (default, 910817479546, 3) exited non-zero on 'Local/8600051@default-581f,1'
[May 15 12:40:42] -- Executing [h@default:1] DeadAGI("Local/8600051@default-581f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----58-----CONGESTION----------") in new stack
[May 15 12:40:42] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 15 12:40:42] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-581f,2'
[May 15 12:40:42] -- Executing [h@default:1] DeadAGI("Local/8600051@default-581f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 15 12:40:42] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 15 12:40:46] == Manager 'sendcron' logged off from 127.0.0.1

Any ideas?

There is always transcoding being done when you're using meetme (and other codecs on your phone).This means that you'll be using more CPU resources when your agents are using g729 on their softphones. If they're on the same LAN then you're better off using Ulaw or Alaw. Use g729 to your VoIP carrier from the GoAutoDial server.

There are a number of reasons why you get voice delays. The most common are network latency and clocksource problems (inaccurate timer). How does your manual call via sofphone compares with manual call via the Vicidial agent UI?

gardo wrote: The most common are network latency and clocksource problems (inaccurate timer). How does your manual call via sofphone compares with manual call via the Vicidial agent UI?

Delays are gone now. The issue i'm facing now is low voice volume on landline side ans a lot of static noise.

Manual call via softphone is ok.

manual call through asterisk on the vicidial server, manual call with the agent logged into a campaign ...? manual call with soft phone straight to provider ...?

if you are dialing straight through or through asterisk without logging in, there would be no transcoding. if those calls are clear, but vicidial is poor quality, you may be overloading your processor with g729. check your server load.

I have been trying to fix my delay problems and thanks to this post i have now got rid of my delays

Thanks guys

im also encountering low voice volume intermittent.

Hi,

This is my first post and I have been scouring the net looking for help and haven't really found much assistance. So I am rather new to Vicidial, and have everything set up but am also experience this audio delay. The quality is clear, and when a call is made directly thru a soft phone of mine with the same carrier there isn't any delay, yet when it's thru the system there is a delay. Could be anywhere from 0.75-2+ seconds. Which of course is a HUGE pain to have that during every conversation. What can I do? I only currently have 3 agents accessing our server remotely now, and the delay will happen with just 1 person, so I know I am not stressing out the server. Any advice? This would mean everything.