[asterisk-users] Asterisk RFC 3261 Compliance

[Sakharam Thorat]

Hello ALL,

Anybody performed ASTERISK Testing for RFC 3261 Compliance?
If Yes,
   Please share Result.

Best Regards,Sakharam Thorat. -- 
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[asterisk-users] Problem with Caller ID when receiving hidden number in via DAHDI and redirecting out via SIP

[Henrik Westerberg]

Hi,

We have a system with both ISDN trunks and SIP. We receive incoming calls on 
both but always dial out via SIP.
When dialing out the caller id is set like this:

exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM})
exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME})
exten => _X.,n,Dial(${CC_DIALSTRING}, 60, em)

This always works fine on SIP and on ISDN as well when the number is not hidden.
But for some reason the setting of the caller id does not work when receiving 
calls from hidden numbers.

The from address in the outgoing SIP looks like this:

From: "Anonymous" 

Does anyone know why this is happening, is there a way to go around it?

Regards,
Henrik

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[asterisk-users] Bus error, Asterisk crash when user leaves a message (ODBC voicemail)

[Hoggins!]

Hello list,

My system behaves in an odd manner, and I can't find why.
When users leave a message on the voicemail, once the message is
recorded and the user hangs up, Asterisk crashes.

I can't figure out when it started to behave like this.

Here is the extract of the dialplan where it occurs :

exten => s,1,Answer()
exten => s,n,Playback(radiom-misenrelation)
exten => s,n,Set(CALLERID(name)=STD RADIOM ${CALLERID(name)})
exten => s,n,Wait(2)
exten => s,n,Dial(IAX2/yomama-out/standard&SIP/standard1,30,mTt)
exten => s,n,Playback(radiom-repondeur)
exten => s,n,VoiceMail(1234@default,s)
exten => s,n,Hangup()

Here is the backtrace PasteBin : http://ur1.ca/fy7nv

It has been generated following the instructions found at :
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Strangely, I'm absolutely sure that the DONT_OPTIMIZE flag is properly
set (I recompiled Asterisk from scratch, making sure this was enabled),
you will still see some "" values.

Should I post this on the asterisk-dev mailing-list, or directly into
the issue tracker ?

Thanks !

Hoggins!

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Re: [asterisk-users] Problem with Caller ID when receiving hidden number in via DAHDI and redirecting out via SIP

[Linus Wiklund]

Hi Henrik.

You might want to read

http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header

and

http://www.voip-info.org/wiki/view/Asterisk+func+callerid


On Mon, Oct 28, 2013 at 11:04 AM, Henrik Westerberg <
henrik.westerb...@ain.se> wrote:

>  Hi,
>
>  We have a system with both ISDN trunks and SIP. We receive incoming
> calls on both but always dial out via SIP.
> When dialing out the caller id is set like this:
>
>  exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM})
> exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME})
> exten => _X.,n,Dial(${CC_DIALSTRING}, 60, em)
>
>  This always works fine on SIP and on ISDN as well when the number is not
> hidden.
> But for some reason the setting of the caller id does not work when
> receiving calls from hidden numbers.
>
>  The from address in the outgoing SIP looks like this:
>
>  From: "Anonymous" 
>
>  Does anyone know why this is happening, is there a way to go around it?
>
>  Regards,
> Henrik
>
>
> --
> _
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>
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Re: [asterisk-users] issue after install dahdi

[Salaheddine Elharit]

Hello

i check the dahdi-channels.conf

in span 1 when i use it like below i can do my outband calls without issue

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 17-31
context = default
group = 63

but when i add in channel 1-15 like: channel => 1-15,17-31

i receive all the time this message

chan_dahdi.c:9438 pri_fixup_principle: Can't move call (DAHDI/3-1) from
channel 3 to 2.  It is already in use.


WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Span 1: PRI
requested channel 1/2 is not available.

could you please help me

thanks and regards







2013/10/24 Salaheddine Elharit 

> ok thanks for your comment i really appreciate it
>
>
> best regards
>
>
> 2013/10/23 Russ Meyerriecks 
>
>> On Wed, Oct 23, 2013 at 11:27 AM, Salaheddine Elharit <
>> salah.elharit...@gmail.com> wrote:
>>
>>> hi
>>>
>>> the issue has been solved after change the span from span
>>> =1,1,0,ccs,hdb3 to span =1,0,0,ccs,hdb3
>>> thanks for everyone
>>>
>>
>> Salaheddine,
>>
>> Just a comment here: I'm not sure who your spans are connected to but, it
>> is highly unlikely that this changed is what fixed your problem. I think
>> it's more likely that the process of reloading something else actually
>> fixed it. What you are doing here is telling span 1 to provide (or ignore)
>> timing to the other end. If it's the case that you're connected to a public
>> e1 pri provider, this probably isn't the correct configuration and will
>> likely cause further problems like slips and alarms. If it's connected to
>> something internal to your business, (like a channel bank), then it's fine.
>>
>> --
>> Russ Meyerriecks
>> Digium, Inc. | Linux Kernel Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> direct: +1 256-428-6025
>> Check us out at: www.digium.com & www.asterisk.org
>>
>> --
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>
>
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[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Eddie Mikell]

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs are
detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine

More than enough bandwidth

Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Put analog lines in the conference room to avoid the dropouts - leave the
sip lines in place for day to day use

Hire a consultant

Ditch the system and buy a pre-packaged system - RingCentral or some such.

There are no local asterisk professionals who can help, and we are a little
leery of opening up our system to outside consultants.

Anyone else face the above, and finally abandoned Asterisk for a commercial
system?

We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.

Suggestions welcome.

Best

Eddie
-- 
Eddie H. Mikell
Senior Systems Engineer
RKG

Office: 434.970.1010 x 124
Email: emik...@rimmkaufman.com

-- 
 
   

    


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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 10/28/2013 01:29:13 PM:

> From: Eddie Mikell 
> To: asterisk-users@lists.digium.com, 
> Date: 10/28/2013 01:29 PM
> Subject: [asterisk-users] Tired of dropouts and garbled phone calls 
> - where to go next?
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> All,
> 
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop 
> outs are detrimental to our sales force.
> 
> I've tried about everything I can think of.  
> 
> Moved the asterisk server from VM machine to dedicated machine
> More than enough bandwidth
> Setting 802.1p = 7
> Set Dedicated voice traffic 35% of bandwidth.
> 
> Not sure what option would be the best
> 
> Put analog lines in the conference room to avoid the dropouts - 
> leave the sip lines in place for day to day use
> Hire a consultant
> Ditch the system and buy a pre-packaged system - RingCentral or some 
such.
> 
> There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
> 
> Anyone else face the above, and finally abandoned Asterisk for a 
> commercial system?  
> 
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
> 
> Suggestions welcome.
> 
> Best
> 
> Eddie

Does the garbled audio and dropouts only occur on outside calls, or do you 
get them on calls between extensions? How is your phone service delivered 
to your site?

If the extension to extension calls are clear, you need to be looking at 
your phone service and how you connect to it. If your local extensions are 
not clear, then you need to look at your asterisk implementation and your 
network.

If your incoming circuits are delivered via SIP, are you sure you have end 
to end QoS between your office and your phone provider? You can set all 
the QoS you want on the packets as they leave your network, but if your 
provider isn't honoring the QoS packets, then you can easily have audio 
issues. One of my locations has SIP service provided over a DSL line from 
the SIP provider. Just last week, the DSL line went down. We routed the 
calls out our standard internet connection and while it did work, we had 
audio dropouts (though only on incoming audio, the other end could hear us 
just fine). As soon as the DSL line was fixed and we routed back over 
their network, all the audio cleared up. It is the difference between low 
ping times and good QoS and higher ping times and providers who may not 
honor QoS.-- 
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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Ron Wheeler]

I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the 
Asterisk - No analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that 
tests pretty close to that most of the time.


I have not found any good tools to track down the causes of poor voice 
quality.

In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my 
voice drops out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls me, I 
call them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so poorly 
on the same environment.


Googling "Asterisk poor audio quality" return several hundred thousand 
references


Ron
On 28/10/2013 2:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone 
service that is being provided.  The choppy phone calls, and drop outs 
are detrimental to our sales force.


I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine

More than enough bandwidth

Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use

Hire a consultant

Ditch the system and buy a pre-packaged system - RingCentral or
some such.

There are no local asterisk professionals who can help, and we are a 
little leery of opening up our system to outside consultants.


Anyone else face the above, and finally abandoned Asterisk for a 
commercial system?


We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the 
conference rooms.


Suggestions welcome.

Best

Eddie
--
Eddie H. Mikell
Senior Systems Engineer
RKG

Office: 434.970.1010 x 124
Email:emik...@rimmkaufman.com 


 
 
 
 








--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mark Wiater]

On 10/28/2013 3:59 PM, Ron Wheeler said:

I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No 
analogue.

I don't have any problems with IAX, but I hear some do.


We have a very lightly loaded 60 Mbs cable link to the Internet that tests 
pretty close to that most of the time.

Bandwidth is less important than the overall quality of the internet link, 
latency and jitter. Either way, there is no QoS on the internet, all bets are 
off.

The codec can matter too. What are you using?



I have not found any good tools to track down the causes of poor voice quality.
In my case, I have good incoming quality and terrible quality going out.

Oh, is your cable connection assymetric? Upload smaller than download? If so, 
that correlates to terrible audio, right?


That is, I can hear people perfectly well but they complain that my voice drops 
out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls me, I call 
them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so poorly on the 
same environment.

Googling "Asterisk poor audio quality" return several hundred thousand 
references

I'd not shoot asterisk yet. I'd focus on the internet connection and it's 
components (cable modem) first.

I use asterisk all over the place. Mostly connected to PRI's and Carrier 
provided SIP trunks, with internet SIP trunks as backup. I get complaints on 
the Internet based SIP trunks sometimes, never on other other two.

I'd ask most of these questions of the OP too. Overall telephony design matters.



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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Eric Wieling]

Does using SIP to your ITSP make any difference?I stopped using IAX2 and 
switched to SIP around 2003 when I experienced similar problems, never looked 
back.  If you insist on using IAX2, then Google for iax2 audio problems

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Monday, October 28, 2013 4:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Tired of dropouts and garbled phone calls - where 
to go next?

I am reaching the same level of frustration.
I have tried to find the source of the problems. 
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No 
analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that tests 
pretty close to that most of the time.

I have not found any good tools to track down the causes of poor voice quality.
In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my voice drops 
out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls me, I call 
them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so poorly on the 
same environment.

Googling "Asterisk poor audio quality" return several hundred thousand 
references 

Ron
On 28/10/2013 2:29 PM, Eddie Mikell wrote:


All, 

The users in our organization are well, quite frankly, sick of phone 
service that is being provided.  The choppy phone calls, and drop outs are 
detrimental to our sales force.

I've tried about everything I can think of.  


Moved the asterisk server from VM machine to dedicated machine

More than enough bandwidth

Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.


Not sure what option would be the best


Put analog lines in the conference room to avoid the dropouts - 
leave the sip lines in place for day to day use

Hire a consultant

Ditch the system and buy a pre-packaged system - RingCentral or 
some such.


There are no local asterisk professionals who can help, and we are a 
little leery of opening up our system to outside consultants.

Anyone else face the above, and finally abandoned Asterisk for a 
commercial system?  

We have 167 users.

I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the 
conference rooms.

Suggestions welcome.

Best

Eddie
-- 

Eddie H. Mikell 
Senior Systems Engineer
RKG

Office: 434.970.1010 x 124
Email: emik...@rimmkaufman.com  

   
    
   
   
    






 




-- 
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mitul Limbani]

Asterisk is a swiss army knife, you should either know how to use it or
rely on ready made software which control routing of calls through variable
bit rates (skype does that very effectively)

So the key here for you to research upon from those several hundred results
is "variable bit rate codec negotiations"

Mitul Limbani
www.facebook.com/enterux
www.facebook.com/entvoice
 On Oct 29, 2013 1:30 AM, "Ron Wheeler" 
wrote:

>  I am reaching the same level of frustration.
> I have tried to find the source of the problems.
> We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk
> - No analogue.
> We have a very lightly loaded 60 Mbs cable link to the Internet that tests
> pretty close to that most of the time.
>
> I have not found any good tools to track down the causes of poor voice
> quality.
> In my case, I have good incoming quality and terrible quality going out.
> That is, I can hear people perfectly well but they complain that my voice
> drops out and is garbled regardless of who places the call.
> As a result,  I use Skype for all of my calls and if someone calls me, I
> call them back on Skype if they have any problems.
> I don't understand why Skype works so well and Asterisk works so poorly on
> the same environment.
>
> Googling "Asterisk poor audio quality" return several hundred thousand
> references
>
> Ron
> On 28/10/2013 2:29 PM, Eddie Mikell wrote:
>
> All,
>
>  The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop outs are
> detrimental to our sales force.
>
>  I've tried about everything I can think of.
>
>   Moved the asterisk server from VM machine to dedicated machine
>
>  More than enough bandwidth
>
>  Setting 802.1p = 7
>
>  Set Dedicated voice traffic 35% of bandwidth.
>
>  Not sure what option would be the best
>
>
>   Put analog lines in the conference room to avoid the dropouts - leave
> the sip lines in place for day to day use
>
>  Hire a consultant
>
>  Ditch the system and buy a pre-packaged system - RingCentral or some
> such.
>
>  There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
>
>  Anyone else face the above, and finally abandoned Asterisk for a
> commercial system?
>
>  We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
>
>  Suggestions welcome.
>
>  Best
>
>  Eddie
> --
> Eddie H. Mikell
> Senior Systems Engineer
> RKG
>
>  Office: 434.970.1010 x 124
> Email: emik...@rimmkaufman.com
>
>  
>    
> 
>     
>
>
>
>
>
>
> --
> Ron Wheeler
> President
> Artifact Software Inc
> email: rwhee...@artifact-software.com
> skype: ronaldmwheeler
> phone: 866-970-2435, ext 102
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mike]

On Mon, 28 Oct 2013, Eddie Mikell wrote:


All,
The users in our organization are well, quite frankly, sick of phone service 
that is being provided.  The choppy phone
calls, and drop outs are detrimental to our sales force.

I've tried about everything I can think of.  

  Moved the asterisk server from VM machine to dedicated machine

  More than enough bandwidth

  Setting 802.1p = 7

  Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best

  Put analog lines in the conference room to avoid the dropouts - leave the 
sip lines in place for day to day use

  Hire a consultant

  Ditch the system and buy a pre-packaged system - RingCentral or some such.

There are no local asterisk professionals who can help, and we are a little 
leery of opening up our system to outside
consultants.

Anyone else face the above, and finally abandoned Asterisk for a commercial 
system?  

We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference 
rooms.

Suggestions welcome.

Best

Eddie
--


As stated in previous replies if you haven't already I would certainly try 
to isolate the problem, e.g., are extension to extension calls good, is 
the problem only on outside calls etc.


We are starting our 4th year of VoIP service and have had two seemingly 
similar episodes to yours during that time.  We are on a non-symmetric 
cable connection, 20/4 (I believe).  After a few days of "crappy" audio I 
started looking for some way to characterize/correlate bad audio with 
something I could measure.  I found iperf (http://iperf.sourceforge.net/) 
to be a free and easy starting point, which actually turned out to be all 
I needed.


I simply ran a "server" instance on our "cloud" server roughly 1K miles 
away and a "client" instance locally.  I used the command line swithces 
that forced udp mode.  This allowed me to see jitter and packet loss in 
both directions.  We had terrible packet loss in the outbound direction. 
This didn't show up in normal browsing, emailing etc., kinds of things as 
I suspect TCP retries masked the problem.  With a little persistence with 
the cable company the second tech found a bad "tap" (I believe) outside at 
the cable drop.  Replacing that solved our issue for almost two years.


The next time this happened iperf showed a similar packet loss problem. 
This time it turned out to be "noise in the system" according to the cable 
tech.  He said it could be from any number of sources but a different team 
would be out to hunt it down the next day.  In the mean time he changed 
out our old Moto SB5101 modem for a more modern DOCSIS 3.0 modem.  The 
multiple channel bonding that it offered was much better at punching 
through the noise.  That change alone ended crappy audio as well as packet 
loss as shown by iperf.-- 
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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mikhail Lischuk]

 

Ron Wheeler писал 28.10.2013 21:59: 

> I have not found any
good tools to track down the causes of poor voice quality.
> In my case,
I have good incoming quality and terrible quality going out.
> That is,
I can hear people perfectly well but they complain that my voice drops
out and is garbled regardless of who places the call.

I had the very
same problems. My users were ready to murder me. 

I've tried whatever I
found relevant, and had no success. ISP was telling me that it was my
problem, everything's ok from their point of view (do they ever say
opposite?). 

As a result, I've changed the ISP and everything became
perfect. I've spent 2 months fighting invisible enemy, but sometimes you
just have to stop trying everything possible and move to impossible
options. 

-- 
With Best Regards
Mikhail
Lischuk

+380681244933

+380504182274

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Duncan Turnbull]

On 29/10/2013, at 9:55 am, Mike  wrote:

> On Mon, 28 Oct 2013, Eddie Mikell wrote:
> 
>> All,
>> The users in our organization are well, quite frankly, sick of phone service 
>> that is being provided.  The choppy phone
>> calls, and drop outs are detrimental to our sales force.
>> I've tried about everything I can think of.  
You probably need to break the problem down into more specific issues.

Does every call suffer or is it just some?
Is the CPU overworked at all and are calls being transcoded or all the same 
codec?
Is it time of day affected?
What load is on the network at the same time?
Do you have monitoring tools such as smoke ping tell you the network looks so 
you can match to issues?
Call drop outs suggest network issues or upset hardware, you shouldn’t be 
getting call drop outs if you have enough bandwidth and low jitter

>> 
>>  Moved the asterisk server from VM machine to dedicated machine
>> 
We have found hardware issues before on hardware and the next box worked fine. 
Testing the network performance to the SIP provider to see jitter and packet 
loss especially while on the phone and getting issues.

>>  More than enough bandwidth
Need both up and down - if you are G711 then its about 128K per channel, if you 
have about 20 simultaneous channels you need about 2M both directions, however 
if its a bandwidth issue you should be fine testing calls when the network load 
is quiet. If you are using GSM etc you need a lot less bandwidth. 
The issue is often the upstream as many services are asymmetric but voip needs 
symmetry. You are sharing with email, cloud apps, vpns and all other stuff so 
make sure there is no congestion on the uplink. 

Buy a link just for voip is often easier than QOS and these days not very 
costly.

>> 
>>  Setting 802.1p = 7
Won’t be honoured on the net
>> 
>>  Set Dedicated voice traffic 35% of bandwidth.

How are you doing this? Mikrotik routers have a good queue mechanism that works 
reasonably well I have found, but the main thing is to test your up and down 
bandwidth and check it stays available under other loads.

>> Not sure what option would be the best
>> 
Get another ISP - or hassle them over performance standards 
>>  Put analog lines in the conference room to avoid the dropouts - leave 
>> the sip lines in place for day to day use
Buy better bandwidth is another option?
Where is your sip provider in relation to your network ? Try another SIP 
provider - the closer and better performing network the better
>> 
>>  Hire a consultant
>> 
>>  Ditch the system and buy a pre-packaged system - RingCentral or some 
>> such.
If you buy a new system and use SIP they will make sure you buy some dedicated 
bandwidth for the network so they don’t have to deal with issues like this

>> There are no local asterisk professionals who can help, and we are a little 
>> leery of opening up our system to outside
>> consultants.
>> Anyone else face the above, and finally abandoned Asterisk for a commercial 
>> system?  
>> We have 167 users.
>> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the 
>> conference rooms.
Well worth getting 10 people dial into conference from desks and check quality 
for 5-10mins - if its good then you are looking at your network

>> Suggestions welcome.

Good luck
>> Best
>> Eddie
>> --
> 


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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Jules Agee]

iperf is great. Another essential troubleshooting tool is nfsen/nfdump 
(or any netflow/sflow monitoring utility that shows DSCP/TOS tag values).


On 10/28/2013 01:55 PM, Mike wrote:
As stated in previous replies if you haven't already I would certainly 
try to isolate the problem, e.g., are extension to extension calls 
good, is the problem only on outside calls etc.


We are starting our 4th year of VoIP service and have had two 
seemingly similar episodes to yours during that time.  We are on a 
non-symmetric cable connection, 20/4 (I believe).  After a few days of 
"crappy" audio I started looking for some way to 
characterize/correlate bad audio with something I could measure. I 
found iperf (http://iperf.sourceforge.net/) to be a free and easy 
starting point, which actually turned out to be all I needed.


I simply ran a "server" instance on our "cloud" server roughly 1K 
miles away and a "client" instance locally.  I used the command line 
swithces that forced udp mode.  This allowed me to see jitter and 
packet loss in both directions.  We had terrible packet loss in the 
outbound direction. This didn't show up in normal browsing, emailing 
etc., kinds of things as I suspect TCP retries masked the problem.  
With a little persistence with the cable company the second tech found 
a bad "tap" (I believe) outside at the cable drop.  Replacing that 
solved our issue for almost two years.


The next time this happened iperf showed a similar packet loss 
problem. This time it turned out to be "noise in the system" according 
to the cable tech.  He said it could be from any number of sources but 
a different team would be out to hunt it down the next day.  In the 
mean time he changed out our old Moto SB5101 modem for a more modern 
DOCSIS 3.0 modem.  The multiple channel bonding that it offered was 
much better at punching through the noise.  That change alone ended 
crappy audio as well as packet loss as shown by iperf.





--
--
Jules Agee
Senior System Administrator
Pacific Coast Feather Co.
jul...@pcf.com   x284

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Patrick Lists]

On 10/28/2013 07:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs
are detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine


That's a good start. Now what have you done to conclude that the 
Asterisk server is not the cause of your problems?



More than enough bandwidth


That's irrelevant. It's about the quality of that bandwidth. Have you 
figured out if there might be a lot of packetloss or are you perhaps on 
a cablelink which is a *shared* medium? Once your link hits the box in 
the street it shares it with others who might be eating up all the 
bandwidth with their torrent downloads etc.? Use tools like iperf, smoke 
ping and mtr to see if there are obvious problems on the route to your 
VoIP provider.



Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Once the packets leave your premises and your ISP/cable company starts 
messing with them a QoS setting is generally not honored so not very 
helpful unless your LAN is congested.



Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use


If those analog lines are cheap, easy to get then as an intermediate 
solution I would order those analog lines as fast as I could. Or fix the 
VoIP problems, whichever is faster.



Hire a consultant


An experienced VoIP consultant should be able to tell you what is or 
could be causing your problems. With your users "sick of phone service" 
it suprises me that you haven't already hired one.



Ditch the system and buy a pre-packaged system - RingCentral or some
such.


And what if it's your Internet link or the route to your VoIP provider? 
What if your VoIP provider is messing up?



There are no local asterisk professionals who can help, and we are a
little leery of opening up our system to outside consultants.


If you don't want that then you don't want that but given the state your 
users are in I would be less worried about giving a Consultant access to 
the Asterisk box and more worried about my job :-)



Anyone else face the above, and finally abandoned Asterisk for a
commercial system?


I have seen that once years ago where some clueless sales guy had 
totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very 
buggy and crash prone. There was no way to make that work reliably. 
After the supplier failed for months I was brought in to review the 
setup and possibly fix it. Told the customer to cut its losses. So they 
kicked out their supplier and opted for a different setup.



We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.


I don't know how Grandstream is these days. I thought the GXP2100 was ok 
but I guess you already know if there's a problem with those phones from 
the (lack of) intra-office call complaints from your users.



Suggestions welcome.


Hire a Consultant or someone who has been part of this Community for a 
while and is well known on this list or in #asterisk on irc. Provide 
remote access if required. Change passwords afterwards.


If you really don't want to provide remote access then find a reputable 
VoIP provider with a switch physically as close as possible to your 
location, get a DID for a few bucks, hook it up to your Asterisk box and 
route it to a line on your phone, grab your cell, call that DID and see 
if you still have the problem. It wouldn't be the first time that the 
link between you and your VoIP provider just doesn't cut it. Or maybe 
your VoIP provider just sucks and you need to change to a different one. 
Both flowroute.com and voip.ms work well for me (no affiliation). Or 
maybe your Internet link sucks and you need to change your ISP.


Good luck.

Regards,
Patrick

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Steve Edwards]

On Mon, 28 Oct 2013, Mike wrote:

I found iperf (http://iperf.sourceforge.net/) to be a free and easy 
starting point, which actually turned out to be all I needed.


I've used iperf to check bandwidth before, but never looked deeper into 
it's features. Thanks for the nudge. Maybe you can help me understand my 
results?


This is how I usually use iperf: (ws is my 'workstation,' kitchen is my 
HTPC)


-kitchen::sedwards:~$ iperf --server

Server listening on TCP port 5001
TCP window size: 85.3 KByte (default)

[  4] local 192.168.0.50 port 5001 connected with 192.168.0.46 port 46692

Client connecting to 192.168.0.46, TCP port 5001
TCP window size:  105 KByte (default)

[  6] local 192.168.0.50 port 49979 connected with 192.168.0.46 port 5001
[ ID] Interval   Transfer Bandwidth
[  6]  0.0-10.0 sec   992 MBytes   832 Mbits/sec
[  4]  0.0-10.0 sec   867 MBytes   726 Mbits/sec

-ws::sedwards:~$ iperf --client kitchen --dualtest

Server listening on TCP port 5001
TCP window size: 85.3 KByte (default)


Client connecting to kitchen, TCP port 5001
TCP window size:  155 KByte (default)

[  5] local 192.168.0.46 port 46692 connected with 192.168.0.50 port 5001
[  4] local 192.168.0.46 port 5001 connected with 192.168.0.50 port 49979
[ ID] Interval   Transfer Bandwidth
[  5]  0.0-10.0 sec   867 MBytes   727 Mbits/sec
[  4]  0.0-10.0 sec   992 MBytes   832 Mbits/sec

Seems reasonable for a 1Gb connection.

Playing with UDP mode, I got some results I wasn't expecting.

-kitchen::sedwards:~$ iperf --server --udp

Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size:  208 KByte (default)

[  3] local 192.168.0.50 port 5001 connected with 192.168.0.46 port 55941

Client connecting to 192.168.0.46, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size:  208 KByte (default)

[  5] local 192.168.0.50 port 47139 connected with 192.168.0.46 port 5001
[ ID] Interval   Transfer BandwidthJitter   Lost/Total Datagrams
[  3]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec   0.036 ms0/  893 (0%)
[  5]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec
[  5] Sent 893 datagrams
[  5] Server Report:
[  5]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec   0.020 ms0/  893 (0%)

-ws::sedwards:~$ iperf --client kitchen --dualtest --udp

Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size:  208 KByte (default)


Client connecting to kitchen, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size:  208 KByte (default)

[  4] local 192.168.0.46 port 55941 connected with 192.168.0.50 port 5001
[  3] local 192.168.0.46 port 5001 connected with 192.168.0.50 port 47139
[ ID] Interval   Transfer Bandwidth
[  4]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec
[  4] Sent 893 datagrams
[  3]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec   0.021 ms0/  893 (0%)
[  4] Server Report:
[  4]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec   0.035 ms0/  893 (0%)

What? Why did my bandwidth dive from 800 Mbits/sec to 1 Mbits/sec?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Doug Lytle]

Steve Edwards wrote:
> What? Why did my bandwidth dive from 800 Mbits/sec to 1 Mbits/sec?

--help shows:

Client specific:
  -b, --bandwidth #[KM]for UDP, bandwidth to send at in bits/sec
   (default 1 Mbit/sec, implies -u)

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Paul Belanger]

On 13-10-28 06:03 PM, Patrick Lists wrote:

On 10/28/2013 07:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs
are detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine


That's a good start. Now what have you done to conclude that the
Asterisk server is not the cause of your problems?


More than enough bandwidth


That's irrelevant. It's about the quality of that bandwidth. Have you
figured out if there might be a lot of packetloss or are you perhaps on
a cablelink which is a *shared* medium? Once your link hits the box in
the street it shares it with others who might be eating up all the
bandwidth with their torrent downloads etc.? Use tools like iperf, smoke
ping and mtr to see if there are obvious problems on the route to your
VoIP provider.


Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Once the packets leave your premises and your ISP/cable company starts
messing with them a QoS setting is generally not honored so not very
helpful unless your LAN is congested.


Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use


If those analog lines are cheap, easy to get then as an intermediate
solution I would order those analog lines as fast as I could. Or fix the
VoIP problems, whichever is faster.


Hire a consultant


An experienced VoIP consultant should be able to tell you what is or
could be causing your problems. With your users "sick of phone service"
it suprises me that you haven't already hired one.


Ditch the system and buy a pre-packaged system - RingCentral or some
such.


And what if it's your Internet link or the route to your VoIP provider?
What if your VoIP provider is messing up?


There are no local asterisk professionals who can help, and we are a
little leery of opening up our system to outside consultants.


If you don't want that then you don't want that but given the state your
users are in I would be less worried about giving a Consultant access to
the Asterisk box and more worried about my job :-)


Anyone else face the above, and finally abandoned Asterisk for a
commercial system?


I have seen that once years ago where some clueless sales guy had
totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very
buggy and crash prone. There was no way to make that work reliably.
After the supplier failed for months I was brought in to review the
setup and possibly fix it. Told the customer to cut its losses. So they
kicked out their supplier and opted for a different setup.


We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.


I don't know how Grandstream is these days. I thought the GXP2100 was ok
but I guess you already know if there's a problem with those phones from
the (lack of) intra-office call complaints from your users.


Suggestions welcome.


Hire a Consultant or someone who has been part of this Community for a
while and is well known on this list or in #asterisk on irc. Provide
remote access if required. Change passwords afterwards.

If you really don't want to provide remote access then find a reputable
VoIP provider with a switch physically as close as possible to your
location, get a DID for a few bucks, hook it up to your Asterisk box and
route it to a line on your phone, grab your cell, call that DID and see
if you still have the problem. It wouldn't be the first time that the
link between you and your VoIP provider just doesn't cut it. Or maybe
your VoIP provider just sucks and you need to change to a different one.
Both flowroute.com and voip.ms work well for me (no affiliation). Or
maybe your Internet link sucks and you need to change your ISP.


^ this

Like others said, you really need to drill down and find out where your 
audio issues are. Local is easy to do, since you control the network, 
remote is harder.


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] Tired of dropouts and garbled phone, calls - where to go next?

[Dave Platt]

> In my case, I have good incoming quality and terrible quality going out.
> That is, I can hear people perfectly well but they complain that my 
> voice drops out and is garbled regardless of who places the call.

This suggests to me that you may have congestion problems in your
"upstream" traffic flow.

Setting QoS on the packets may not help, if whatever router you are
using for Internet connectivity isn't managing the traffic flow well.
In my experience, you need to do two things:

-  Make sure that the traffic with QoS for low latency, is placed
   in a separate transmission queue on the router from "bulk" traffic
   (e.g. web service, file transfer).

-  Make sure that your router uses a "traffic shaping" system, to
   ensure that data isn't being submitted to the network interface
   faster than it can actually be transmitted by the *slowest*
   link in the path to your Internet provider.

A lot of routers, switches, and network interface drivers these days
have a lot of buffering.  This is a mixed blessing.  Unless you are
careful, a burst of low-priority (bulk) traffic can be transmitted
into your switch/router, and fill up a bunch of the buffer... by the
time the system "knows" that you have some audio-QoS traffic to send,
there's a whole bunch of data ahead of it in some network router or
switch (or even the ring-buffer in a network interface card) and there's
no way for your audio data to "jump ahead" of the bulk data in order to
be delivered quickly.

Ideally, your system should send data upstream at a rate which never
"bursts" up to, or above your link's sustained data transmission rate.
You want the buffers in the "upstream" equipment to remain as empty
as possible.

For background reading on this, look up the "Bufferbloat" problem
and project, and the "Linux ultimate traffic shaper" scripts.
They may not be directly applicable to your problem but may
explain some of what you are seeing.



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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Ron Wheeler]

On 28/10/2013 4:12 PM, Mark Wiater wrote:


On 10/28/2013 3:59 PM, Ron Wheeler said:

I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the 
Asterisk - No analogue.

I don't have any problems with IAX, but I hear some do.

I have now switched to SIP and will check the quality in the morning.


We have a very lightly loaded 60 Mbs cable link to the Internet that 
tests pretty close to that most of the time.
Bandwidth is less important than the overall quality of the internet 
link, latency and jitter. Either way, there is no QoS on the internet, 
all bets are off.


The codec can matter too. What are you using?

G711




I have not found any good tools to track down the causes of poor 
voice quality.

In my case, I have good incoming quality and terrible quality going out.
Oh, is your cable connection assymetric? Upload smaller than download? 
If so, that correlates to terrible audio, right?

Just ran a test 50 Mbps  download 10Mbps upload. Should be enough I hope.


That is, I can hear people perfectly well but they complain that my 
voice drops out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls 
me, I call them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so 
poorly on the same environment.


Googling "Asterisk poor audio quality" return several hundred 
thousand references
I'd not shoot asterisk yet. I'd focus on the internet connection and 
it's components (cable modem) first.


Good idea. I am sure that you are right but what to test and how are not 
clear.
I use asterisk all over the place. Mostly connected to PRI's and 
Carrier provided SIP trunks, with internet SIP trunks as backup. I get 
complaints on the Internet based SIP trunks sometimes, never on other 
other two.


I'd ask most of these questions of the OP too. Overall telephony 
design matters.







--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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