vicidial.org

Hello there, As many of you has read in other posts, I've a very big issue with outbound calls.
I really can't find the origin of this bad thing, It's my first experiment with vicidial so I really don't know where to look. I know that AVG load has to be low, not too much calls on the same machine, but we're working max with 8 agents and only one campaign with RATIO dial 2.0.

The major issues:
- Delayed (5-6 secs.) calls
- Agents can hear the customer while customer doesn't, and viceversa
- echoes
- If I listen a conference I can hear both, but they really make hard to talk each other.
- Sometimes when they logout and relogin the things become better (for a while)
- When they call directly from the softphones (Express Talk) the calls goes like a charme



I hope this is my last post about this issues, I really don't want to "spam" the forum with duplicate posts!

I'll write here my configuration, hopefully, if you need something else to know, please tell me!


Hardware:
DB+WEB+ASTERISK
Asus P5QPRO
CPU Intel Q9800
4 GB RAM DDR2 800
4x 80 GB 15.000 RPM SAS Hard Disks in RAID 1+0 on 3ware 9750 controller
1x SIP trunk (24 channels ulaw (or G729) on 2 Mbit/s SHDSL)
Timing provided by Sangoma Voicetime USB (AVG 99,994%)
Firewall Linksys WRT54GL with tomato firmware (QOS enabled)

Software:
Vicibox Redux downgraded to asterisk 1.4.21 and zaptel (same issue with dahdi and asterisk 1.4.27)
astguiclient 2.2.0 RC6

Load:
Max 8 agents only outbound with no recording at all
2 Campaigns with 30.000 leads each (not used simultaneously)

some data from CLI:

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vicidialDBWEB:~ # cat /proc/interrupts
           CPU0       CPU1       CPU2       CPU3
  0:         75          1          0          9   IO-APIC-edge      timer
  1:          2          4          2          2   IO-APIC-edge      i8042
  8:          0          0          1          0   IO-APIC-edge      rtc0
  9:          0          0          0          0   IO-APIC-fasteoi   acpi
 16:      28445       1884        450        451   IO-APIC-fasteoi   3w-sas, uhci_hcd:usb3
 18:          0          0          0          0   IO-APIC-fasteoi   ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8
 19:        130         23       4181       4159   IO-APIC-fasteoi   uhci_hcd:usb7, ata_piix, ata_piix
 21:          0          0          0          0   IO-APIC-fasteoi   uhci_hcd:usb4
 23:      52705      12943    3169902    3136748   IO-APIC-fasteoi   ehci_hcd:usb2, uhci_hcd:usb6
217:      67929         13         13         11   PCI-MSI-edge      eth1
218:          1          0          0          0   PCI-MSI-edge      eth0
NMI:          0          0          0          0   Non-maskable interrupts
LOC:     419720     224697     109854     184070   Local timer interrupts
RES:       2667       2324       2120       1829   Rescheduling interrupts
CAL:        611        639        615        126   function call interrupts
TLB:       3484      10215       2973      10052   TLB shootdowns
TRM:          0          0          0          0   Thermal event interrupts
SPU:          0          0          0          0   Spurious interrupts
ERR:          0
MIS:          0


Today's report:


Our SIP trunk:

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vicidialDBWEB*CLI> sip show peer Alida
vicidialDBWEB*CLI>

  * Name       : Alida
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-pstn
  Subscr.Cont. : <Not set>
  Language     : en
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : 0287365384
  FromDomain   : sip.alida.it
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Nat          : Always
  ACL          : Yes
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : auto
  LastMsg      : 0
  ToHost       : sip.alida.it
  Addr->IP     : 194.244.4.155 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: 0287365384
  SIP Options  : (none)
  Codecs       : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status       : OK (31 ms)
  Useragent    :
  Reg. Contact :


How many SIP carriers have you tried?

what codecs have you tried?

Just a guess, which sound card you have in agent pc ??

Realtek ?

Timming issue?

have you run zttest -v

please post that result

Did you try installing from scratch?

@Matt: Tried only with this carrier (G729, uLAW, ALAW and GSM) We can't use other carriers: in Italy is very hard to find a business grade voip carrier that offers more than 3 calls simultaneously on the same trunk. We're thinking to switch to a T1 (E1 here) line, but we have to wait 'till the end of the contract (1 year).

@Michael_N: Timing issue? I really don't hope so! I've tried X100P, Sangoma A102 and now, reading your results and happyness with it, I'm using a Sangoma Voicetime! Here's the test:

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--- Results after 180 passes ---
Best: 100.000 -- Worst: 99.991 -- Average: 99.998063, Difference: 99.999745

@boybawang: No, I've installed from ViciboxRedux, had a lot of problem with the raid controller driver. They're pretty new and only supports OpenSuse or kernels > .33
The installation script of viciboxredux is not like installing from scratch?

@gmcust3:
Do you mean that Realtek is not compatible with Vicidial? :O
Yes, is Realtek AC97!
Here's the whole client config:
Windows XP Pro SP3
No firewall enabled
AMD Sempron 3400+
1 GB Ram
Mozilla Firefox 3.0 or 3.5 or 3.6

What kind of network connection do you use?

have you tried using only alaw/ulaw codecs both on clientphones and on trunk?

Are you recording your calls?

can you do a sip show peers and do a mtr from your server to your voice provider to check if there are packet losses along the way:

mtr ip.address.of.your.voip.provider

also did you try using zoiper or bria or eyebeam, and disabled Acoustic Echo Cancel , Auto Gain Control, ANd noise Reduction

did you try calling your extensions, like your on cc100 then call cc110 to test the quality

Michael_N wrote:What kind of network connection do you use?

have you tried using only alaw/ulaw codecs both on clientphones and on trunk?

Are you recording your calls?

Thank you for answering. Do you mean LAN connection? 24 ports 100 Mbit switch uplinked with a 1 Gbit switch where all server are connected. This served us for more than 2 years flawless, with up than 40 calls on the same lan with no issues (using directly softphones of course)

Right now we're using uLAW both for voip trunk and client phones, with no recording.

how many agents are logged on when this happen?

Is there difference between manual dial and autodail?

boybawang wrote:can you do a sip show peers and do a mtr from your server to your voice provider to check if there are packet losses along the way:

mtr ip.address.of.your.voip.provider

also did you try using zoiper or bria or eyebeam, and disabled Acoustic Echo Cancel , Auto Gain Control, ANd noise Reduction

did you try calling your extensions, like your on cc100 then call cc110 to test the quality

I've tried with Express Talk and Xlite so far, today I'll test the other clients. But AEC, AGC and NR are already disabled on the actual softphones.

You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Also try USB based headsets for better audio quality. I suggest plantronics DSP 400 or higher

Michael_N wrote:how many agents are logged on when this happen?

Is there difference between manual dial and autodail?

Happens with more than 2-3 agents connected, we never used manual dial so far, only adaptive_average (6%) and ratio (2.0)

Have you tried it on ratio 1.0 ?

You could try add some more ram

Or move the mysql server to another computer

Michael_N wrote:You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Done points 2, 3 and 4. I can't do points 6 and 7. But with so low load will this change significally the performance?. I've posted it mainly for very large implementations

Michael_N wrote:Have you tried it on ratio 1.0 ?

You could try add some more ram

Or move the mysql server to another computer

Actually we have 8 GB RAM installed on it, but it obviously sees only just about 4 GB. I'm planning to move Asterisk on another machine in these days.

DarknessBBB wrote:

Michael_N wrote:You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Done points 2, 3 and 4. I can't do points 6 and 7. But with so low load will this change significally the performance?. I've posted it mainly for very large implementations

I havent looked at it, but it might have changed the priority mysql is running at , so that mysql uses more "power" so that asterisk gets bad calls quality.

You shoud do a clean install and what happening

Michael_N wrote:

DarknessBBB wrote:

Michael_N wrote:You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Done points 2, 3 and 4. I can't do points 6 and 7. But with so low load will this change significally the performance?. I've posted it mainly for very large implementations

I havent looked at it, but it might have changed the priority mysql is running at , so that mysql uses more "power" so that asterisk gets bad calls quality.

You shoud do a clean install and what happening

I wanted to say that... NOW I've done those points

DarknessBBB wrote:

Michael_N wrote:

DarknessBBB wrote:

Michael_N wrote:You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Done points 2, 3 and 4. I can't do points 6 and 7. But with so low load will this change significally the performance?. I've posted it mainly for very large implementations

I havent looked at it, but it might have changed the priority mysql is running at , so that mysql uses more "power" so that asterisk gets bad calls quality.

You shoud do a clean install and what happening

I wanted to say that... NOW I've done those points

I meant that doing those points could have influance on other software running at the computer..

Michael_N wrote:

DarknessBBB wrote:

Michael_N wrote:

DarknessBBB wrote:

Michael_N wrote:You posted this link

http://www.randombugs.com/linux/tuning- ... abase.html

did you do what they recommended?

Done points 2, 3 and 4. I can't do points 6 and 7. But with so low load will this change significally the performance?. I've posted it mainly for very large implementations

I havent looked at it, but it might have changed the priority mysql is running at , so that mysql uses more "power" so that asterisk gets bad calls quality.

You shoud do a clean install and what happening

I wanted to say that... NOW I've done those points

I meant that doing those points could have influance on other software running at the computer..

I understand you, but I didn't do those points so far.

We faced the issue regarding Realtek..

May be , just a try , u can try Yamaha chip..

Also if your using realtek disable the software mixer that came with it, did you try calling extension to extension?

boybawang wrote:Also if your using realtek disable the software mixer that came with it, did you try calling extension to extension?

The Realtek mixes is disabled, and internal direct calls are good.

Maybe can this help?

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WARNING[3143]: chan_sip.c:13675 handle_response: Remote host can't match request NOTIFY to call '4fbb7afa5cba3208202e765373f015e0@192.168.1.10'. Giving up.

Take a look at my post in this tread

http://www.vicidial.org/VICIDIALforum/v ... hp?t=10695

Thank you for answering Michael. I've read it carefully and it looks interesting. But if it would be the problem everyone here would have this issue, am I right?

since your calls is working fine when do a direct call with your softphone

But as soon you use vicidial and invoke meetme then the problem starts am i right?

Michael_N wrote:since your calls is working fine when do a direct call with your softphone

But as soon you use vicidial and invoke meetme then the problem starts am i right?

Yep, also "auto-dial campaigns to dial without any live agents and play a message" work fine with up to 22 concurrent calls...

DarknessBBB wrote:

Michael_N wrote:since your calls is working fine when do a direct call with your softphone

But as soon you use vicidial and invoke meetme then the problem starts am i right?

Yep, also "auto-dial campaigns to dial without any live agents and play a message" work fine with up to 22 concurrent calls...

I'm thinking along the same lines as Michael_N here. Try run this command at the asterisk shell

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zap show status

It should say wanpipe_voicetime for the Sangoma usb timing device. If it says ztdummy, that's the problem.

Thank you for replying James. Here's the output:

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viciAST*CLI> dahdi show status
Description                              Alarms     IRQ        bpviol     CRC4 
WANVTIME/1 (source: wanpipe_voicetime)   UNCONFIGUR 0          0          0 

Hi DarknessBBB

That result disproves my idea. It show that asterisk is correctly using the timing device. When using a timing device asterisk conferences should remain in sync.

If I come up with another idea I'll let you know.

DarknessBBB , have u been able to solve the Issue of RealTek ?

I started using Yamaha or ESS chip but not getting these chips.

So, switched to USB but call volume is really very low.

Need Solution !!

Hello there, I figured out that the issue was related to our VoIP provider. Since we switched from VoIP to E1 lines, the quality is absolutely flawless.
Meanwhile I've learned that using a Sangoma USB Timer and USB based headphones (ie. Sennheiser) can improve the overall volume and quality of the conversation.

I tried USB.Its Ok but Not very Loud.

I use 4 simultaneous Providers and Issue with all.

I cant switch to E1 as I do International Calls.

I heard RealTek Sound Mixer Creates the MajoR ISSUE.

But How to remove it ?