RE: [Asterisk-Users] Asterisk's MultiProcessor Ability

2005-05-26 Thread Gary Lawrence 
I use a Dual Xeon hyper threaded, top shows it as 4 cpus and the load seems
to be pretty well balanced.

Sincerely;

Gary Lawrence
ITcom.Net
866.4ITcom1
866.448.2661

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Wednesday, May 25, 2005 12:40 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk's MultiProcessor Ability

We have asterisk running on a quad processor dell. The kernel has been
compiled with SMP.

However, asterisk seems to only use 1 processor. 3 of the 4 always stay at
100% idle.

Is it pointless to have a multi-proc machine? I was going to buy a new dual
3.6Ghz Xeon server but if nothing will take advantage of the other proc...

Perhaps my conception of multi-proc/threaded is warped. If asterisk is the
only thing using CPU, I would expect the load to be distributed amounst the
processors. Instead of 1 proc falling to 20% idle (80% using on that 1
proc), I should see all 4 procs fall to 80% idle (20% used on each). Is this
wrong?

What about the g729 library from digium? Is that multi-proc aware?

-Matthew

-- 

Matthew Boehm, IT DirectorCypress Telecommunications
[EMAIL PROTECTED]   3838 N. Sam Houston Parkway E #400
T: 832-200-8640 x3044  Houston, TX 77032

My girlfriend was recently diagnosed with multiple personality disorder;
 When she called yesterday, my CallerID box exploded.


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RE: [Asterisk-Users] IAX-IAX Trunking not works

2005-05-25 Thread Gary Lawrence 
Trunking works for me. I'm not sure what the problem is but can have you try
different things till we find it.

Notransfer=yes doesn't work for me. Calls still transfer.

Try putting trunk=yes in EVERY user.

Also I don't use type=friend. Try setting up a seperate user and peer
context.

Sincerely;

Gary Lawrence
ITcom.Net

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adnan Ahmed
Sent: Wednesday, May 25, 2005 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX-IAX Trunking not works

no man iax2 trunking not working i don't know why its really odd 
iax2 trunk debug command shows
IAX2 Trunk Debug Requested
Beginning trunk processing
Ending trunk processing with 0 peers and 0 calls processed
wat's that means how can i enable trunking on one ser iax2 show
channels command shows:
asteriskser1*CLI> iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq
(Tx/Rx)  Lag  Jitter  JitBuf  Format
IAX2/[EMAIL PROTECTED]  192.168.0.151test2   3/3 
00121/00108  00020ms  0006ms  0056ms  gsm
IAX2/[EMAIL PROTECTED]  192.168.0.151test2   6/8 
6/3  00013ms  0001ms  0049ms  gsm
on another server shows

test2*CLI> iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq
(Tx/Rx)  Lag  Jitter  JitBuf  Format
IAX2/[EMAIL PROTECTED]/2192.168.0.77 adnan   2/20687 
00026/00023  [Native Bridged to ID=4]
IAX2/192.168.0.51:45  192.168.0.51 test2   4/4 
00021/00025  [Native Bridged to ID=2]
IAX2/[EMAIL PROTECTED]/5  192.168.0.79 iphone  5/25617 
00025/00024  [Native Bridged to ID=6]
IAX2/192.168.0.51:45  192.168.0.51 test2   6/3 
00021/00026  [Native Bridged to ID=5]
4 active IAX channel(s)
is something going wrong plz i am very keen to solve this as soon as
possible plz kindly enlighten on this issue.

> 
> IAX2 Trunk Debug Requested
> Beginning trunk processing
> Ending trunk processing with 1 peers and 3 calls processed
> 
> If you want to free up more bandwidth add "echocancel=no" to your iax.conf
> 
> Gary Lawrence
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve Clark
> Sent: Monday, May 23, 2005 10:01 AM
> To: Adnan Ahmed; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IAX-IAX Trunking not works
> 
> Adnan Ahmed wrote:
> > Hello ,
> > I want some tips guidance i am sure this topic discuss alot in list,i
> > try my best to solve it by myself try googling looking wiki everywhere
> > but no luck question is iax-iax trunking not working setting,trying
> > each n every option
> >
> > server2 iax.conf:
> > [general]
> > bindport=4569
> > bandwidth=low
> > disallow=all
> > allow=gsm
> > jitterbuffer=no
> > tos=lowdelay
> > trunk=yes
> > notransfer=yes
> >
> > [saim]
> > username=saim
> > secret=saim
> >
> > type=friend
> > host=dynamic
> > context=from-sip
> >
> > disallow=all
> > allow=gsm
> >
> > [noman]
> > username=saim
> > secret=noman
> > type=friend
> > host=dynamic
> > context=from-sip
> > disallow=all
> > allow=gsm
> >
> > [asteriskser1]
> > type=friend
> > ;auth=md5
> > ;secret=qwerty
> > context=local
> > ;host=dynamic
> > defaultip=192.168.0.51
> > notransfer=yes
> > qualify=no
> > trunk=yes
> > canreinvite=no
> >
> > server1 iax.conf:
> > [general]
> > bindport=4569
> > bandwidth=low
> > disallow=all
> > allow=gsm
> > jitterbuffer=no
> > tos=lowdelay
> > trunk=yes
> > notransfer=yes
> >
> > [user1]
> > username=user1
> > secret=user1
> > type=friend
> > host=dynamic
> > context=from-sip
> > disallow=all
> > allow=gsm
> >
> > [user2]
> > username=user2
> > secret=user2
> > type=friend
> > host=dynamic
> > context=from-sip
> > disallow=all
> > allow=gsm
> >
> > [test2]
> > type=friend
> > context=local
> > defaultip=192.168.0.51
> > notransfer=yes
> > qualify=no
> > trunk=yes
> > canreinvite=no
> >
> >
> > I am using Kiax soft phone  on both servers using codec GSM asterisk
> > latest stable version OS SLES9 ,any help is highly appreciated i had
> > look almost every place in wiki regarding iax trunking but all in
> > vein.
> > Thanks In Advance.
> > ___
>

RE: [Asterisk-Users] IAX-IAX Trunking not works

2005-05-24 Thread Gary Lawrence 
While you have active calls, type at the cli prompt "iax2 trunk debug".

If trunking is working you should get a reply like:

IAX2 Trunk Debug Requested
Beginning trunk processing
Ending trunk processing with 1 peers and 3 calls processed

If you want to free up more bandwidth add "echocancel=no" to your iax.conf

Gary Lawrence

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Clark
Sent: Monday, May 23, 2005 10:01 AM
To: Adnan Ahmed; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX-IAX Trunking not works

Adnan Ahmed wrote:
> Hello ,
> I want some tips guidance i am sure this topic discuss alot in list,i
> try my best to solve it by myself try googling looking wiki everywhere
> but no luck question is iax-iax trunking not working setting,trying
> each n every option
> 
> server2 iax.conf:
> [general]
> bindport=4569
> bandwidth=low
> disallow=all
> allow=gsm
> jitterbuffer=no
> tos=lowdelay
> trunk=yes
> notransfer=yes
> 
> [saim]
> username=saim
> secret=saim
> 
> type=friend
> host=dynamic
> context=from-sip
> 
> disallow=all
> allow=gsm
> 
> [noman]
> username=saim
> secret=noman
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [asteriskser1]
> type=friend
> ;auth=md5
> ;secret=qwerty
> context=local
> ;host=dynamic
> defaultip=192.168.0.51
> notransfer=yes
> qualify=no
> trunk=yes
> canreinvite=no
> 
> server1 iax.conf:
> [general]
> bindport=4569
> bandwidth=low
> disallow=all
> allow=gsm
> jitterbuffer=no
> tos=lowdelay
> trunk=yes
> notransfer=yes
> 
> [user1]
> username=user1
> secret=user1
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [user2]
> username=user2
> secret=user2
> type=friend
> host=dynamic
> context=from-sip
> disallow=all
> allow=gsm
> 
> [test2]
> type=friend
> context=local
> defaultip=192.168.0.51
> notransfer=yes
> qualify=no
> trunk=yes
> canreinvite=no
> 
> 
> I am using Kiax soft phone  on both servers using codec GSM asterisk
> latest stable version OS SLES9 ,any help is highly appreciated i had
> look almost every place in wiki regarding iax trunking but all in
> vein.
> Thanks In Advance.
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
Please don't use reply when you are starting a new thread.
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RE: [Asterisk-Users] G729 codec

2005-05-22 Thread Gary Lawrence 









At the cli prompt type “show codecs”.
In the right hand column it states G.729A.



Sincerely; 

Gary Lawrence 
ITcom.Net 
866.4ITcom1

866.448.2661




-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of todd
Sent: Sunday, May 22, 2005 12:55
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] G729
codec 

 



Hi all
I have a question  and hope it has not been answered before. I have
searched the forums and mail but have not seen this answered conclusively.
Does the G729 codec and licenses which digium sales for asterisk use g729 a
or b or both; I have had a hard time getting a conclusive answer.





If it does use g729b how could I show evidence to
a client that it is b and not a?
Thanks

Sincerely
TKG








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RE: [Asterisk-Users] having asterisk play music on holdtocallerswhile phone rings?

2005-05-21 Thread Gary Lawrence 









Yours could look totally different than
mine depending on how you route calls.

 

It will start with “exten” and
have the word “Dial” in it. You may have several lines that you
need to change...

 

In the below example change the r at the
end to an m.

 

exten => _NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}||r



 

 

 

 

yep





I have hold music other wise





looks like I am going to have to go
in to the [EMAIL PROTECTED] and configure it via
that method





can you give me pointers on what the
dial line looks like so I dont screw this thing up??





they dont recommend editing this
stuff bye hand unless you know what you are doing.





thanks





hank





 





email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5







- Original Message - 





From: Gary Lawrence 





To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' 





Sent: Saturday,
May 21, 2005 2:09 PM





Subject: RE:
[Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings?





 



Edit the extensions.conf
and put an m at the end of the dial line.



Do you have hold music
otherwise?

Sincerely; 

Gary Lawrence 
ITcom.Net 
866.4ITcom1

866.448.2661




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday, May 21, 2005 4:26
PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] having
asterisk play music on hold to callerswhile phone rings?

 



hello how do I set up asterisk to
play music on hold to callers while it rings my  phones?





I am using the amp portal to
configure the asterisk pbx just to let you all know.





thanks





hank





 





email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5









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RE: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings?

2005-05-21 Thread Gary Lawrence 









Edit the extensions.conf and put an m at
the end of the dial line.



Do you have hold music otherwise?

Sincerely; 

Gary Lawrence 
ITcom.Net 
866.4ITcom1

866.448.2661




-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday, May 21, 2005 4:26
PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] having
asterisk play music on hold to callerswhile phone rings?

 



hello how do I set up asterisk to
play music on hold to callers while it rings my  phones?





I am using the amp portal to
configure the asterisk pbx just to let you all know.





thanks





hank





 





email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5








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