[asterisk-users] sip trunking and call transfer

[nik600]

Hi to all.

i-ve got a question:

what happen when a call between 2 trunks is transferred to another trunk?

For example, suppose that i have 4 trunk A,B,C,D:

Caller 1 - Trunk A/B - Caller2

Then Caller 2 transfer to Caller 3 behind Trunk B/C

What happend?

a) Caller 1 - Trunk A/B - Trunk B/C - Caller3

or

b) Caller 1 - Trunk A/C - Caller3

So:

is it possible to avoid the scenario a) ?

Thanks to all
-- 
/*/
nik600
http://www.kumbe.it

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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Al Baker]

Dan Austin wrote:
> Yehavi wrote:
>   
>>  Our university has to upgrade soon its old Nortel PBX's
>> which holds around 10,000 extensions tied to 5 PBXes. Up
>> to now we thought about commercial solutions but now
>> there is a window openning for open source solution.
>> However, I need examples to convince that this solution
>> is feasible, and preferably at other universities.
>> 
>
>   
>> Are there any pointers for such installations?
>> 
>
> Sam Houston University migrated from a Cisco CallManager
> and Nortel setup to Asterisk a couple years back.
>
> I do not know any of the specific details, but maybe
> you can track down someone involved in the project.
>
> Dan
>
>   
Remember - You are going from a CARRIER GRADE purpose built piece of 
hardware with Software built under a rigid CMM with extensive 
"soak-testing" to software that has been developed under , shall we say, 
a somewhat less rigid and stringent methodology.
You will be moving from an environment supported by hundreds of highly 
trained people, some with decades of TELCO experience
to one where you support comes from a somewhat less seasoned group of 
individuals.
10,000 extensions ???
On Asterisk ???
You pays your money, you takes you chances.
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>   

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[asterisk-users] ISDN for Dahdi get hangs

[Clive.Chan(ATN)]

Hi all, 
I am using TE220 on Centos 4.7 final, asterisk 1.4.22, Dahdi 2.0.0, and
everytime get hangs after I type the  "dadhi_cfg -vv" and what is my
problem? Can some one help me for this?
 
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Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

Some are mis dialed but most work one day but not the next
they are all dialed manually

Robb

Don Kelly wrote:
> What is the source of the numbers you are calling? Are they
> previously-verified numbers from your database? Are some of them
> fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more
> than 3% of calls that I manually call. Have you researched some of the
> failures (examining the numbers that were attempted to be called)? I don't
> really see a problem with what you're reporting.
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office TM
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>  
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Robert
> Boardman
> Sent: Thursday, November 20, 2008 4:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] ISDN Cause codes
>
> Hi All
>
> Just been looking at stats for one of my sites, and I'm conserned about 
> the number of error cause codes being returned from the telco
>
> for example
>
> 12000 calls processed
>
> 131 are cause code 31* normal. unspecified.*
>
> 139 are cause code 28 * invalid number format (address incomplete).*
>
> 112 are cause code 1 *Unallocated (unassigned) number.
>
> *this adds up to about 3% of calls not completing.
>
> there are various other codes including 17 busy 34 channel unavaliable 
> and 44 requested channel unavaliable, which add up to another 1%.*
> *
> the telco says there is no problem with the line, I'm trying to 
> understand what the problem could be
>
> now  alot of calls complete OK so I don't think is my configs
>
> Any advice would be appriciated
>
> Versions
> asterisk 1.4.21.1
> zaptel 1.4.12.1
>
>
> Robb
>
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Dan Austin]

Yehavi wrote:
>  Our university has to upgrade soon its old Nortel PBX's
> which holds around 10,000 extensions tied to 5 PBXes. Up
> to now we thought about commercial solutions but now
> there is a window openning for open source solution.
> However, I need examples to convince that this solution
> is feasible, and preferably at other universities.

> Are there any pointers for such installations?

Sam Houston University migrated from a Cisco CallManager
and Nortel setup to Asterisk a couple years back.

I do not know any of the specific details, but maybe
you can track down someone involved in the project.

Dan

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[asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Yehavi Bourvine]

Hello,

  Our university has to upgrade soon its old Nortel PBX's which holds around
10,000 extensions tied to 5 PBXes. Up to now we thought about commercial
solutions but now there is a window openning for open source solution.
However, I need examples to convince that this solution is feasible, and
preferably at other universities.

Are there any pointers for such installations?

   Thanks! __Yehavi:
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[asterisk-users] Group count not being preserved when transferring a call into a conference

[Daniel Johnson]

Hi,

I am using Group and Group_Count to limit the number of calls to go out
over a single peer as our channels with that peer is limited to 8.

If we dial and outside number over this peer and then transfer the call
into a MeetMe conference the Group gets decremented when it should not?
This is most likely an error on my behalf, however I am not sure what
the correct solution is.

I have set the MeetMe conference up on a local extention 777.

exten => 777,1,Answer 
exten => 777,n,MeetMe(9003|rpM)  
exten => 777,n,Playback(vm-goodbye)  
exten => 777,n,Hangup

Do I need to do something in the above to preserve the Group count?

Also there have been some complaints about callee's phone line being
tied up and connected to the conference even after they hangup? Does
meeetme not detect hangups?
These calls have gone out over an IAX2 peer, is there something special
I must do?

Regards

-- 

Daniel
Johnson
Systems Administrator / Systems Development
Scanning Systems Australia




Office: +61 7 3387 
Facsimile: +61 7 3387 5588
E-mail: [EMAIL PROTECTED]
Website: http://www.scanningsystems.com.au




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Re: [asterisk-users] Full Duplex

[Adam Lovegrove]

I have the same issue, using a TE121 with hardware echo cancellation.
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Re: [asterisk-users] A way to run extenrnotify when IMAP events take place...

[Jeffrey Phelps]

But how do I get it to run a script??  I don't have any SMDI Interfaces,
so I wouldn't be able to put anything in the config...

 

Thanks,

 

Jeff

 

Jeffrey Phelps schrieb:

> I have IMAP voicemail working with Exchange 2003 using a single
username

> and password for multiple mailboxes.

 

> Right now, I am setting up asterisk to use voicemail with my Cisco
Call

> Manager (Which I detest BTW...) and I have everything working, EXCEPT:

 

> I cannot get my externnotify script to run when any changes have been

> made to the VoiceMail...

 

> Scenario:

 

> Bob gets a call  -> Bob rejects call to voicemail

 

> Caller leaves Bob a voicemail  -> externnotify calls script which
turns

> on his Cisco MWI.

 

> Bob checks Voicemail  ->  Bob deletes Voicemail  -> asterisk says that

> the voicemail was deleted, but doesn't run my script again to turn off

> the Cisco MWI.

 

> I would just like to know if there is any work around for this.

> OR.  Maybe Someone is working on adding this into the code

> so that it works...

 

> I'm running * 1.6.1-beta2

 

afaicr I read something which might be related in doc/smdi.txt.

 

 

   Philipp Kempgen

 

 

 

Thanks,

 

Jeff Phelps

IT Support Specialist

 

McConnell Jones Lanier and Murphy, LLP  

3040 Post Oak Blvd., Suite 1600, Houston, TX 77056

(713) 968-1600 (phone)

(713) 968-1688 (direct phone)

(713) 968-1601 (main fax)

http://www.mjlm.com/  

 

IRS Circular 230 Disclosure: To ensure compliance with requirements
imposed by the IRS, McConnell & Jones, LLP informs you that any U.S.
federal tax advice contained in this communication (including any
attachments, enclosures, or other accompanying material) is not intended
or written to be used, and cannot be used, for the purpose of (i)
avoiding penalties under the Internal Revenue Code or (ii) promoting,
marketing, or recommending to another party any transaction or matter
addressed herein; for IRS audit, tax disputes or other purposes.

 

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Re: [asterisk-users] Full Duplex

[Matt Riddell]

On 18/11/2008 9:46 a.m., Matthew Fredrickson wrote:
> Singer X.J. Wang wrote:
>> We've had the same issue. For calls that go between a SIP connection 
>> (desktop phones) and Zaptel connections, there was a lot of problems 
>> with half duplex. We switched
>> from the Digium card to the Sangoma card and the problem went away.
> 
> Just for the record, he said that it happened regardless of protocol (IP 
> to IP calls do not use the card based echo cancellers).
> 
> Sorry for the problem you had.  However, I think that if you use a 
> current version of our echo canceller board, you will find your issues 
> resolved.  In fact, for a significant number of Digium's boards, Sangoma 
> uses the exact same hardware echo canceller.

A few months ago, I had a similar problem and needed to pass:

vpmnlptype=4 vpmnlpmaxsupp=11

to resolve it. If I upgraded zaptel would this be fixed?

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Steve Totaro]

On Thu, Nov 20, 2008 at 6:27 PM, Dave Platt <[EMAIL PROTECTED]> wrote:
>> Coming from outside the network, setting up for a couple rounds of
>> NATting isn't going to work well.  They are not seeing it between
>> phones.  Others, using the polycom phones have reported echo between two
>> SIP on a 4ms ping trip.
>
> Could this be due to a purely acoustic echo within the Polycom handsets?
>
> I encountered a nasty echo / hollow sound when using a cheap USB
> "telephone" to connect to my Asterisk system (via KPhoneSI).  The
> echoing was due to acoustic feedback - the handset body acted as a
> very nice channel for sound waves from the back side of the
> speaker down to the microphone cartridge.
>
> I opened up the handset, added some damping materials (panel-
> vibration-damping and soft-foam sheeting, left over from a
> car stereo speaker installation I did), closed it back up,
> and the echoing was gone.
>
> You might not notice in some calls, if the Polycom phones have
> silence-detection turned on for those calls and if the amount
> of feedback falls below the phones' silence threshold.  If
> the phone silence-detection algorithm were turned off on
> other calls, the echo would then be audible.
>

Troubleshooting is simple.  Register X-Lite and see if the problem
goes away.  If so it is IAX2, a moving target that sometimes works OK
but more often does not.

If the X-Lite softphone does not show echo, then it has something to
do with the Polycoms,  If you still have echo, then drop IAX2, if that
doesn't do it, then mess with the polycoms.

When will Asterisk get VAD?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] SVN - Digium

[Luis Morales]

Thnx Philipp!!



On Fri, Nov 21, 2008 at 7:12 PM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> Luis Morales schrieb:
>> Does any know what happens with svn repository on svn.digium.com ?
>
> http://lists.digium.com/pipermail/asterisk-users/2008-November/222147.html
>
>
>   Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
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>



-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] Playback using AMI

[Jim Dickenson]

I am not sure how one would do that. If I have a dialplan that does a
chanspy the dialplan hangs on that step so how could you do a background
step?

I am clearly missing something in your suggestion.

I am using version 1.6.0.1 if that makes a difference.

Here is more about my setup.

One leg of the call is created by a logged in agent being added to a queue.
While the other leg of the call is the result of an originate action that
queues the call when it is answered.

I use the monitor action to record either or both legs of the call.

I use an originate action to a third leg that executes a dialplan that does
a chanspy to one of the active legs to "spy" on the call.

What do you suggest I use to play a sound file so that the original two
legs, or all three legs if there is spying going on, can hear it?

The monitor, spy and playing of sound files are all independent things that
I might want to perform with the original two legs.
-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/



> From: Danny Nicholas <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Thu, 20 Nov 2008 16:46:54 -0600
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: Re: [asterisk-users] Playback using AMI
> 
> Just set up a new "spy" in the dialplan that performs a Background on the
> sound file, then hangs up.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
> Sent: Thursday, November 20, 2008 4:34 PM
> To: Asterisk User MailList
> Subject: [asterisk-users] Playback using AMI
> 
> Is there a way to inject sound from a sound file into an established call
> using AMI?
> 
> I have an established call from which I can record either or both legs. I
> can additionally "spy" on the call. Is there any way I can play a sound file
> into the call and not loose the ability for the people to continue talking
> while listening to the sound file?
> -- 
> Jim Dickenson
> mailto:[EMAIL PROTECTED]
> 
> CfMC
> http://www.cfmc.com/
> 
> 
> 
> 
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Re: [asterisk-users] SVN - Digium

[Philipp Kempgen]

Luis Morales schrieb:
> Does any know what happens with svn repository on svn.digium.com ?

http://lists.digium.com/pipermail/asterisk-users/2008-November/222147.html


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] A way to run extenrnotify when IMAP events take place...

[Philipp Kempgen]

Jeffrey Phelps schrieb:
> I have IMAP voicemail working with Exchange 2003 using a single username
> and password for multiple mailboxes.

> Right now, I am setting up asterisk to use voicemail with my Cisco Call
> Manager (Which I detest BTW...) and I have everything working, EXCEPT:

> I cannot get my externnotify script to run when any changes have been
> made to the VoiceMail...

> Scenario:

> Bob gets a call  -> Bob rejects call to voicemail

> Caller leaves Bob a voicemail  -> externnotify calls script which turns
> on his Cisco MWI.

> Bob checks Voicemail  ->  Bob deletes Voicemail  -> asterisk says that
> the voicemail was deleted, but doesn't run my script again to turn off
> the Cisco MWI.

> I would just like to know if there is any work around for this.
> OR.  Maybe Someone is working on adding this into the code
> so that it works...

> I'm running * 1.6.1-beta2

afaicr I read something which might be related in doc/smdi.txt.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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[asterisk-users] SVN - DIGIUM

[Luis Morales]

Does any know what happens with svn repository on svn.digium.com ?

-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Dave Platt]

> Coming from outside the network, setting up for a couple rounds of
> NATting isn't going to work well.  They are not seeing it between
> phones.  Others, using the polycom phones have reported echo between two
> SIP on a 4ms ping trip.

Could this be due to a purely acoustic echo within the Polycom handsets?

I encountered a nasty echo / hollow sound when using a cheap USB
"telephone" to connect to my Asterisk system (via KPhoneSI).  The
echoing was due to acoustic feedback - the handset body acted as a
very nice channel for sound waves from the back side of the
speaker down to the microphone cartridge.

I opened up the handset, added some damping materials (panel-
vibration-damping and soft-foam sheeting, left over from a
car stereo speaker installation I did), closed it back up,
and the echoing was gone.

You might not notice in some calls, if the Polycom phones have
silence-detection turned on for those calls and if the amount
of feedback falls below the phones' silence threshold.  If
the phone silence-detection algorithm were turned off on
other calls, the echo would then be audible.


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Re: [asterisk-users] ISDN Cause codes

[Don Kelly]

What is the source of the numbers you are calling? Are they
previously-verified numbers from your database? Are some of them
fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more
than 3% of calls that I manually call. Have you researched some of the
failures (examining the numbers that were attempted to be called)? I don't
really see a problem with what you're reporting.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2008 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes

Hi All

Just been looking at stats for one of my sites, and I'm conserned about 
the number of error cause codes being returned from the telco

for example

12000 calls processed

131 are cause code 31* normal. unspecified.*

139 are cause code 28 * invalid number format (address incomplete).*

112 are cause code 1 *Unallocated (unassigned) number.

*this adds up to about 3% of calls not completing.

there are various other codes including 17 busy 34 channel unavaliable 
and 44 requested channel unavaliable, which add up to another 1%.*
*
the telco says there is no problem with the line, I'm trying to 
understand what the problem could be

now  alot of calls complete OK so I don't think is my configs

Any advice would be appriciated

Versions
asterisk 1.4.21.1
zaptel 1.4.12.1


Robb

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[asterisk-users] A way to run extenrnotify when IMAP events take place...

[Jeffrey Phelps]

I have IMAP voicemail working with Exchange 2003 using a single username
and password for multiple mailboxes.

 

Right now, I am setting up asterisk to use voicemail with my Cisco Call
Manager (Which I detest BTW...) and I have everything working, EXCEPT:

 

I cannot get my externnotify script to run when any changes have been
made to the VoiceMail...

 

Scenario:

 

Bob gets a call  -> Bob rejects call to voicemail

 

Caller leaves Bob a voicemail  -> externnotify calls script which turns
on his Cisco MWI.

 

Bob checks Voicemail  ->  Bob deletes Voicemail  -> asterisk says that
the voicemail was deleted, but doesn't run my script again to turn off
the Cisco MWI.

 

 

I would just like to know if there is any work around for this.
OR.  Maybe Someone is working on adding this into the code
so that it works...

 

I'm running * 1.6.1-beta2

 

Any help is appreciated, and thanks in advance.

 

Thanks,

 

Jeff Phelps

IT Support Specialist

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[asterisk-users] ISDN Cause codes

[Robert Boardman]

Hi All

Just been looking at stats for one of my sites, and I'm conserned about 
the number of error cause codes being returned from the telco

for example

12000 calls processed

131 are cause code 31* normal. unspecified.*

139 are cause code 28 * invalid number format (address incomplete).*

112 are cause code 1 *Unallocated (unassigned) number.

*this adds up to about 3% of calls not completing.

there are various other codes including 17 busy 34 channel unavaliable 
and 44 requested channel unavaliable, which add up to another 1%.*
*
the telco says there is no problem with the line, I'm trying to 
understand what the problem could be

now  alot of calls complete OK so I don't think is my configs

Any advice would be appriciated

Versions
asterisk 1.4.21.1
zaptel 1.4.12.1


Robb

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Re: [asterisk-users] Playback using AMI

[Danny Nicholas]

Just set up a new "spy" in the dialplan that performs a Background on the
sound file, then hangs up.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
Sent: Thursday, November 20, 2008 4:34 PM
To: Asterisk User MailList
Subject: [asterisk-users] Playback using AMI

Is there a way to inject sound from a sound file into an established call
using AMI?

I have an established call from which I can record either or both legs. I
can additionally "spy" on the call. Is there any way I can play a sound file
into the call and not loose the ability for the people to continue talking
while listening to the sound file?
-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/




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Re: [asterisk-users] Limit the number of users in a meetme conference?

[Dan Austin]

Noah wrote:
> I found the "maxusers" defined in meetme.c, but I'm
> not sure how this value is set.  Does anybody know
> if one can limit the number of users permitted in a
> meetme conference?  I know there's MeetmeCount(), but
> I'd rather avoid the dialplan logic and just set
> maxusers instead.

That feature is primarily used with RealTime conferences.
The maxusers value is read from a database and enforced
on RealTime enable conferences.  This presumes you are
looking at 1.6.X or Trunk code...

Dan

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Re: [asterisk-users] Limit the number of users in a meetme conference?

[Singer X.J. Wang]

Be careful there, the size of a int is not statically defined in c.  The 
rules in the standard that applies to an int is as follows:


An |int| must not be larger than a |long int|
A |long int| must be at least 32 bits long int
An |int| must be at least 16 bits long. 

So an int is at least 16 bits (thus max value of 2^15-1) but there is no 
upper limit. I've found that in most modern systems and compilers; I've 
done a fair bit of programming

and in most modern systems, ints are 32 bits (max value of 2^31-1)

Danny Nicholas wrote:

In my meetme.c, users is defined as an int on line 328.  This gives a
possibility of 35768 people in a conference.  If you cbanged that to a
signed char, you would limit it to 127.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Thursday, November 20, 2008 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Limit the number of users in a meetme conference?

Hi -

I found the "maxusers" defined in meetme.c, but I'm not sure how this
value is set.  Does anybody know if one can limit the number of users
permitted in a meetme conference?  I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set maxusers instead.


Thanks,
Noah

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--
*Singer Wang*
/System and Database Engineer/
The Pythian Group

Office: (613) 565-8696 x298
Toll Free:  (877) 798-4426 x298
Fax:(613) 565-8710
Email:  [EMAIL PROTECTED]
MSN:[EMAIL PROTECTED]
Yahoo:  pythianwang
AIM:pythianwang
ICQ:201253
Gadu-Gadu:  6817795
Tencent QQ: 858310404

begin:vcard
fn:Singer Wang
n:Wang;Singer
email;internet:[EMAIL PROTECTED]
tel;work:(613) 565-8696 x298
x-mozilla-html:TRUE
version:2.1
end:vcard

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[asterisk-users] Playback using AMI

[Jim Dickenson]

Is there a way to inject sound from a sound file into an established call
using AMI?

I have an established call from which I can record either or both legs. I
can additionally "spy" on the call. Is there any way I can play a sound file
into the call and not loose the ability for the people to continue talking
while listening to the sound file?
-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/




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[asterisk-users] OT: ATA causes random DTMF in stream

[OCG Technical Support]

I've got a user with Linksys ATA's for their analog phones.  At random times
during calls, the other party hears DTMF tones during the call.

 

Is there a way to solve this?

 

 

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Re: [asterisk-users] Limit the number of users in a meetme conference?

[Danny Nicholas]

In my meetme.c, users is defined as an int on line 328.  This gives a
possibility of 35768 people in a conference.  If you cbanged that to a
signed char, you would limit it to 127.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Thursday, November 20, 2008 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Limit the number of users in a meetme conference?

Hi -

I found the "maxusers" defined in meetme.c, but I'm not sure how this
value is set.  Does anybody know if one can limit the number of users
permitted in a meetme conference?  I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set maxusers instead.


Thanks,
Noah

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[asterisk-users] Limit the number of users in a meetme conference?

[Noah Miller]

Hi -

I found the "maxusers" defined in meetme.c, but I'm not sure how this
value is set.  Does anybody know if one can limit the number of users
permitted in a meetme conference?  I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set maxusers instead.


Thanks,
Noah

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[c james]

Steve Totaro wrote:
> On Thu, Nov 20, 2008 at 1:13 PM, c james <[EMAIL PROTECTED]> wrote:
>> Steve Totaro wrote:
>>> Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
>>> IAX2 is not all it is cracked up to be.
>>>
>>> Also, do a ping to see latency,  200ms is pretty much my standard.
>>>
>> Coming from outside the network, setting up for a couple rounds of
>> NATting isn't going to work well.  They are not seeing it between
>> phones.  Others, using the polycom phones have reported echo between two
>> SIP on a 4ms ping trip.
>>
> 
> NAT is manageable with OpenVPN and very easy.  You just need a box on
> both sides.
> 
> Also, a more difficult setup will allow SIP to work through NAT if
> both sides are behind a NAT.  I just prefer OpenVPN because it is set
> it and forget it.
> 
> Anyways, it is quite simple to switch to SIP to test.  IAX2 has made
> me quite a bit of money because of it's "issues", where SIP "Just
> Works"
> 


I'll get the network guards involved and see.


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Re: [asterisk-users] SIP to IAX2 with delayed echo

[c james]

Drew Gibson wrote:
> c james wrote:
>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>> having a conversation.  Call quality is reported as good except for an
>> echo with a 3 second delay.
>>
>> Most of my searches are saying echo happens only on the PSTN piece, but
>> there isn't one here.
>>
>>   
> 
> Which end hears the echo?
> 
> If it is the Polycom end, try a better quality headset with the softphone.
> Echo comes from analogue portions of the "circuit" and is usually caused 
> at the end that doesn't hear it.
> 
> regards,
> 
> Drew
> 

Both side are seeing it.


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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Benny Amorsen]

c james <[EMAIL PROTECTED]> writes:

> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
> having a conversation.  Call quality is reported as good except for an
> echo with a 3 second delay.

Feedback from speaker to microphone. The problem is always at the end
which doesn't hear it.


/Benny


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[asterisk-users] DTMF issue

[michel freiha]

Hi all,

Kindly note that I got the below message when sending DTMF in RFC2833
through asterisk PBX...The DTMF is not going through

RTCP Read too short

I'm using G729 codec and asteriks Asterisk 1.4.21.2

Regards
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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Tim Panton]

Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause  
echo ?

Tim.

On 20 Nov 2008, at 18:47, Steve Totaro wrote:

> Simple tests.  Change from the highly touted "IAX2" to SIP, but before
> that, download X-Lite and see if you have the same delay.  If you
> don't then look at your Polycoms, if you do, then switch to SIP.
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
>
> On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton
> <[EMAIL PROTECTED]> wrote:
>> There are also settings which will turn on local echo cancellation  
>> for
>> the handset, headset and/or speaker phone. I don't recall their  
>> names at
>> the moment. They are off by default on the handset and headset unless
>> you're using a very recent (3.0+) SIP app.
>>
>> Tim Nelson wrote:
>>> I'm not sure about the 3 second delay, but I've seen plenty of  
>>> echo issues on Polycom phones when the gain has been changed on  
>>> the handset. Check the voice.gain.tx and voice.gain.rx settings in  
>>> your sip.cfg to make sure they're not too high.
>>>
>>> You also may want to make sure there aren't any system resource  
>>> constraints such as high CPU usage or memory usage... :-)
>>>
>>> Tim Nelson
>>> Systems/Network Support
>>> Rockbochs Inc.
>>> (218)727-4332 x105
>>>
>>> - "c james" <[EMAIL PROTECTED]> wrote:
>>>
 A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
 having a conversation.  Call quality is reported as good except for
 an
 echo with a 3 second delay.

 Most of my searches are saying echo happens only on the PSTN piece,
 but
 there isn't one here.

 Can someone point me in the right direction?

 Asterisk 1.4.21.2
 Under 40 users
 Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's  
 what
 they wanted to use!)
>
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[asterisk-users] dial console/dsp hear crackling in headset

[Jerry Geis]

I am calling the console dsp and speaking just fine.
however, I am hearing crackling and all kinds of static.
I go into alsamixer and mute the MIC channel, bring the levels to zero
and nothing affects it.

Why might that be? I expected to not hear anything - especially after 
muting the channel and levels to zero.

Jerry

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[asterisk-users] Elastix workshop in Toronto; Wed Nov 26th, 2008

[Simon P. Ditner]

This Wednesday, November 26th, the Toronto Asterisk Users Group invites 
all in the area to join us for a telephony workshop and talk sponsored by 
Sangoma Inc.[1]

Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix, 
will be running a "getting started" workshop on Elastix, followed by a 
talk discussing how it differs from other Asterisk-based distributions, 
and a road map of the project's future.

Elastix[3] is an open source asterisk-based linux telephony appliance 
that integrates tools such as OpenFire IM Server, SugarCRM, mail 
services, and billing software into a single, easy-to-use interface. It 
also adds its own set of utilities and allows for the creation of third 
party modules.

When:
  Wednesday November 26th, 2008
  5:00 pm - 7:00 pm: WORKSHOP - Getting Started with Elastix (reg. req.)
  7:00 pm - 8:00 pm: TALK - Integrated Communications with Elastix

Where:
  Committee Room 3
  North York Civic Centre (in Mel Lastman Square)
  5100 Yonge St.,
  North York, ON
  Map link: http://xrl.us/hqbw

Registration is requested for the workshop, sign up at: http://taug.ca/node/174
No registration is required for the talk: http://taug.ca/node/175

Check back at http://taug.ca for event updates.

Cheers,
Simon P. Ditner
TAUG.ca Talk Coordinator

[1] http://www.sangoma.com
[2] http://www.palosanto.com
[3] http://www.elastix.org


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Re: [asterisk-users] A question about how much an Asterisk Dcap consultant and a Sipmaster make

[Steve Totaro]

I would say little if not none.  I always give my clients a
satisfaction guaranteel and if they are any good at google, they find
me or get referrals.  I actually do not do any marketing and stay
busy.  I cannot comment on Sip Master.

I think it is more of a status thing in the community at this point.

Anyways, six or seven years of four FXO boxes to a DS3 15k calls a day
speaks or itself.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

On Thu, Nov 20, 2008 at 3:14 PM, Scott Berry <[EMAIL PROTECTED]> wrote:
> Hello there,
>
> I am wondering if some one could tell me on the average in the U.S. What
> does a person with DCap certification make on a standard Asterisk
> installation and configuration process as well as a Sip Master. I am looking
> to go to the Asterisk course and I am blind and have a state agency possibly
> paying for my training and I would like to find out what the average wages
> are so that I can put this in my report. Thanks for any help.
>
> Have a wonderful day.
> Scott Berry
> [EMAIL PROTECTED]
\

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[asterisk-users] A question about how much an Asterisk Dcap consultant and a Sipmaster make

[Scott Berry]

Hello there,

I am wondering if some one could tell me on the average in the U.S. What does a 
person with DCap certification make on a standard Asterisk installation and 
configuration process as well as a Sip Master.  I am looking to go to the 
Asterisk course and I am blind and have a state agency possibly paying for my 
training and I would like to find out what the average wages are so that I can 
put this in my report.  Thanks for any help.

Have a wonderful day.
Scott Berry
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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Philipp Kempgen]

Eric "ManxPower" Wieling schrieb:
> Philipp Kempgen wrote:
>> Olivier schrieb:
>> 
>>> For a long time, I was wondering if I should use MAC address instead of
>>> Extension number to identify SIP endpoints (as I'm mostly not using
>>> softphones).
>>>
>>> Before diving into this, I wondered how people using MAC address are using
>>> CLI as it seems more natural and simple to type
>>> "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
>>> Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ?
>> 
>> SIP accounts are about users (not extensions or devices).
> 
> No, SIP "accounts" are about devices.  There is nothing in sip.conf 
> (except maybe the callerid= settings) that tie the information to a 
> specific user.

I would rather put it like this:
There is nothing in sip.conf that should tie the information to a
specific device.  :-)

> In fact the same user can use multiple devices, all 
> listed in sip.conf.

True. Or:
A human user can use multiple SIP accounts (be they on different
devices or just on one).
Just as he/she can use multiple email/Jabber/... accounts.
That's why email addresses like [EMAIL PROTECTED] are so uncommon.
Remember that a user agent is primarily meant to act on behalf
of a user.

That might be nit-picking for many installations but it's important
when you want mobility / hot-desking.

But in the end the peer entries are just names and thus seem to
be a matter of taste. I won't argue about it.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Drew Gibson]

c james wrote:
> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
> having a conversation.  Call quality is reported as good except for an
> echo with a 3 second delay.
>
> Most of my searches are saying echo happens only on the PSTN piece, but
> there isn't one here.
>
>   

Which end hears the echo?

If it is the Polycom end, try a better quality headset with the softphone.
Echo comes from analogue portions of the "circuit" and is usually caused 
at the end that doesn't hear it.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] Rv: sas

[sas sas]

- Mensaje reenviado 
De: sas sas <[EMAIL PROTECTED]>
Para: asterisk-users@lists.digium.com
Enviado: miércoles, 5 de noviembre, 2008 9:20:34
Asunto: sas


I would like to know how can I send and receive sms using asterisk 1.4.22.
If anyone know, please tell me soon.
Thanks.


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[asterisk-users] VB 6 developer needed

[Gregory Malsack]

Hello Everyone,

 

I’ve sold an asterisk system to a client that has a custom written CRM package 
written in VB6 with an MS-SQL backend. They want to “unplug” the application 
from their old phone system and “plug” it into the asterisk system. The program 
has pop-up screens based on incoming calls. Buttons to re-direct calls 
(transfer, hold, voicemail) on the pop-up screen. The app can also initiate 
calls. I would also need someone who can send CDR information to their MS-SQL 
database. I’ve been told that if asterisk dumps it’s CDR to postgres, triggers 
in postgres can send that information to MS-SQL via ODBC in some fashion.

 

Any help on this would be greatly appreciated!


Greg


No virus found in this outgoing message.
Checked by AVG. 
Version: 7.5.549 / Virus Database: 270.9.8/1801 - Release Date: 11/20/2008 9:11 
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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Eric "ManxPower" Wieling]

Olivier wrote:
> 2008/11/20 Eric ManxPower Wieling <[EMAIL PROTECTED]>
> 
>> Personally I use the MAC-x wherex=the line appearance number.  MAC-a for
>> first line appearance, MAC-b for 2nd, etc.
>>
> 
> Is it easy to use (CLI, logs...) ?
> Would you step back to an extension-based identification scheme ?

I would never go back to extension based sip accounts.

Some of our users have multiple different extensions per phone and many 
extensions appear on more than one phone.  It is very nice to be able to 
say to the user "read me stuff on the white sticker on the bottom of the 
phone" (which is where Polycom puts the MAC).  Then we know EXACTLY 
which phone the user is talking about, which phone config file, which 
parts of the dial plan, etc.


-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Steve Totaro]

Simple tests.  Change from the highly touted "IAX2" to SIP, but before
that, download X-Lite and see if you have the same delay.  If you
don't then look at your Polycoms, if you do, then switch to SIP.
-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)


On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton
<[EMAIL PROTECTED]> wrote:
> There are also settings which will turn on local echo cancellation for
> the handset, headset and/or speaker phone. I don't recall their names at
> the moment. They are off by default on the handset and headset unless
> you're using a very recent (3.0+) SIP app.
>
> Tim Nelson wrote:
>> I'm not sure about the 3 second delay, but I've seen plenty of echo issues 
>> on Polycom phones when the gain has been changed on the handset. Check the 
>> voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure 
>> they're not too high.
>>
>> You also may want to make sure there aren't any system resource constraints 
>> such as high CPU usage or memory usage... :-)
>>
>> Tim Nelson
>> Systems/Network Support
>> Rockbochs Inc.
>> (218)727-4332 x105
>>
>> - "c james" <[EMAIL PROTECTED]> wrote:
>>
>>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>>> having a conversation.  Call quality is reported as good except for
>>> an
>>> echo with a 3 second delay.
>>>
>>> Most of my searches are saying echo happens only on the PSTN piece,
>>> but
>>> there isn't one here.
>>>
>>> Can someone point me in the right direction?
>>>
>>> Asterisk 1.4.21.2
>>> Under 40 users
>>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
>>> they wanted to use!)

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Dave Fullerton]

There are also settings which will turn on local echo cancellation for 
the handset, headset and/or speaker phone. I don't recall their names at 
the moment. They are off by default on the handset and headset unless 
you're using a very recent (3.0+) SIP app.

Tim Nelson wrote:
> I'm not sure about the 3 second delay, but I've seen plenty of echo issues on 
> Polycom phones when the gain has been changed on the handset. Check the 
> voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're 
> not too high.
> 
> You also may want to make sure there aren't any system resource constraints 
> such as high CPU usage or memory usage... :-)
> 
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
> 
> - "c james" <[EMAIL PROTECTED]> wrote:
> 
>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>> having a conversation.  Call quality is reported as good except for
>> an
>> echo with a 3 second delay.
>>
>> Most of my searches are saying echo happens only on the PSTN piece,
>> but
>> there isn't one here.
>>
>> Can someone point me in the right direction?
>>
>> Asterisk 1.4.21.2
>> Under 40 users
>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
>> they wanted to use!)
>>
>>
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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

[Steve Totaro]

On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
>> 2008/11/17 Philipp Kempgen <[EMAIL PROTECTED]>
>>
>> > Tilghman Lesher schrieb:
>> > > On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
>> > >> Is there somewhere a statement from Digium how long they will support
>> > >> Asterisk 1.4?
>> > >
0>> > > There is no statement, because we haven't even discussed when
the EOL for
>> > > 1.4 will be reached.  Certainly that means it won't happen for at least
>> > the
>> > > next 60 days, but beyond that, I really don't know.
>> >
>> > For the average non-techie user who does not want to compile
>> > themselves that may sound funny (if not scary).
>> >
>> > When Debian Lenny (featuring Asterisk 1.4) is finally going to be
>> > released that version might not even be supported any more.
>>
>>
>> I think to a large extend, Asterisk is not to be considered as binary
>> distributed at all, as many hardware it supports is not directly managed by
>> kernel team.
>
> Interesting consideration. Debian Etch and RHEL5 are based on kernel
> 2.6.18, but support quite a few hardware devices not included in that
> kernel.
>
> If this issue bothers you, please help test the alternative timing
> mechanism support now included in trunk.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>

I still compile and install 1.2 for the most part, for call centers
and large systems.

The few 1.4 installs that I have done have been for "medium" sized
PBXs, say 50-70 phones/users and they have been trouble free for the
most part.  Safe_asterisk may make some troubles transparent.

I am not really sure what 1.4 has over 1.2 for the average PBX installation.

Then you have the OpenPBX guys who forked 1.2, I know they have added
functionality to 1.2, but the following puts me off.  Perhaps
vaporware, perhaps not, it all relies on the devs.  You also have
people like Matt Florell who have continued to add functionality to
1.2 but since Digium won't take them, or the dev doesn't want to sign
over their first born, they are hard to come by but certainly out
there.

1.4 may follow the same path, being forked.

1.6 is not on my radar.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Steve Totaro]

On Thu, Nov 20, 2008 at 1:13 PM, c james <[EMAIL PROTECTED]> wrote:
> Steve Totaro wrote:
>> Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
>> IAX2 is not all it is cracked up to be.
>>
>> Also, do a ping to see latency,  200ms is pretty much my standard.
>>
>
> Coming from outside the network, setting up for a couple rounds of
> NATting isn't going to work well.  They are not seeing it between
> phones.  Others, using the polycom phones have reported echo between two
> SIP on a 4ms ping trip.
>

NAT is manageable with OpenVPN and very easy.  You just need a box on
both sides.

Also, a more difficult setup will allow SIP to work through NAT if
both sides are behind a NAT.  I just prefer OpenVPN because it is set
it and forget it.

Anyways, it is quite simple to switch to SIP to test.  IAX2 has made
me quite a bit of money because of it's "issues", where SIP "Just
Works"

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[c james]

Steve Totaro wrote:
> Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
> IAX2 is not all it is cracked up to be.
> 
> Also, do a ping to see latency,  200ms is pretty much my standard.
> 

Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work well.  They are not seeing it between
phones.  Others, using the polycom phones have reported echo between two
SIP on a 4ms ping trip.


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Re: [asterisk-users] SIP to IAX2 with delayed echo

[c james]

Tim Nelson wrote:
> I'm not sure about the 3 second delay, but I've seen plenty of echo issues on 
> Polycom phones when the gain has been changed on the handset. Check the 
> voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're 
> not too high.
> 
> You also may want to make sure there aren't any system resource constraints 
> such as high CPU usage or memory usage... :-)
> 
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
> 
> - "c james" <[EMAIL PROTECTED]> wrote:
> 
>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>> having a conversation.  Call quality is reported as good except for
>> an
>> echo with a 3 second delay.
>>
>> Most of my searches are saying echo happens only on the PSTN piece,
>> but
>> there isn't one here.
>>
>> Can someone point me in the right direction?
>>
>> Asterisk 1.4.21.2
>> Under 40 users
>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
>> they wanted to use!)

Gains are at their default values.  Definitely no problem with the
resources.


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[asterisk-users] Subversion Mirror Down for Maintenance

[Russell Bryant]

Greetings,

We recently moved our public subversion mirror to a new server.  It is 
currently down for maintenance while we resolve some unforeseen 
problems.  It should be back up by the end of the day.

I apologize for the inconvenience,

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Olivier]

2008/11/20 Eric ManxPower Wieling <[EMAIL PROTECTED]>

>
> Personally I use the MAC-x wherex=the line appearance number.  MAC-a for
> first line appearance, MAC-b for 2nd, etc.
>

Is it easy to use (CLI, logs...) ?
Would you step back to an extension-based identification scheme ?
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[asterisk-users] Disable native bridge?

[Tod Fitch]

Background:
WAN1 - Fixed IP low latency, low jitter
WAN2 - Fixed IP medium latency, higher jitter than I like for good VoIP
Firewall/Router not SIP aware
NATed LAN
Asterisk on server located on LAN.
Most, but not all ATA/IP phones on LAN

In the past I was running a v1.2 Asterisk which acted as a B2BUA (all  
RTP streams relayed through Asterisk server) thus presenting only one  
SIP device to the FW. Used rules in the FW to allow SIP and RTP to/ 
from Asterisk and WAN1. Used manual FW and Asterisk (external IP)  
reconfigure for failover to less desirable WAN2 if WAN1 failed.


Upgraded to Asterisk 1.4 and now internal ATA/IP phones are now  
attempting to send RTP streams directly to the Internet. Amazingly,  
this seems to work for my primary ITSP (I wonder what magic they are  
using to map RTP datagrams from a different IP/port than the SIP setup  
negotiated?). But it does not work for ENUM destinations.


I have tried various sip.conf changes (nat=yes/no, canreinvite=yes/no/ 
nonat and directrtpsetup=yes/no) values trying to get all RTP traffic  
to go through the Asterisk box instead of direct but have been unable  
to do so. Any suggestions?


I know, the best way would be to get a SIP aware FW but replacing the  
current one is not in the budget nor is there an old computer sitting  
around that is suitable to press into service as a FW.


--Tod



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[asterisk-users] Low RX volume and half duplex/"walkie-talkie" on AEX-804E

[Lincoln King-Cliby]

Hi All,

I have a ticket open with Digium, but based on their previous lack of support 
for the Asterisk Appliance, I'm not really holding my breath - and, honestly, 
I'm not 100% convinced it's a Digium issue in the first place (but I don't know 
where else to point fingers).

We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a 
Dell PowerEdge 1950 with four straight analog telephone lines, and running 
asterisk 1.4.22. All of the local phones are Cisco 7961G with the SIP firmware. 
Calls between SIP sets, across our SIP trunk on a VPN to a remote office, or 
calls to or from the remote office's PSTN lines (over the aforementioned SIP 
trunk) are all fine.

On many [but not all] calls to or from the PSTN, I'm getting two complaints -
#1 is low receive (i.e. from the PSTN) volume
#2 (which seems to get significantly worse if I try tweaking bumping up the 
tx/rx gain in Zapata.conf) is that if the person in our office is talking all 
inbound audio is muted, but not the other way around (i.e. half duplex, but not 
half duplex both directions if that makes any sense)

Further compromising my sanity is that #1 seems hard for me to duplicate - 
calls to or from my cell phone, for example, always sound fine. Local calls are 
"mostly" fine, and long distance calls are hit-or-miss, calls to a Hawaiian 
(how's that for "Long Distance" from Ohio) 1004 Hz test number are fine - in 
fact, subjectively, borderline too loud which makes no sense since before going 
live with Asterisk, we had a legacy Panasonic KSU/PBX on the same lines - on 
the same punchdown blocks - and no one ever complained about these issues.

If I turn off the echo canceller there's a modest (may even just be 
psychological) improvement in line gain, but the echo is so horrendous 
(actually the echo sounds louder than the inbound call volume) as to make 
things unusable.

Any ideas? At all? I'm still relatively new to the 
Asterisk-interconnected-to-PSTN side of things, and it seems like there are 
dozens of config files and tools so explicit instructions are appreciated!

Thanks in advance,

Lincoln

--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/
Crestron Authorized Independent Programmer

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Eric "ManxPower" Wieling]

Philipp Kempgen wrote:
> Olivier schrieb:
> 
>> For a long time, I was wondering if I should use MAC address instead of
>> Extension number to identify SIP endpoints (as I'm mostly not using
>> softphones).
>>
>> Before diving into this, I wondered how people using MAC address are using
>> CLI as it seems more natural and simple to type
>> "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
>> Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ?
> 
> SIP accounts are about users (not extensions or devices).

No, SIP "accounts" are about devices.  There is nothing in sip.conf 
(except maybe the callerid= settings) that tie the information to a 
specific user.  In fact the same user can use multiple devices, all 
listed in sip.conf.

Personally I use the MAC-x wherex=the line appearance number.  MAC-a for 
first line appearance, MAC-b for 2nd, etc.

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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[asterisk-users] Sending / Receiving sms messages with Portech 370

[Julian Lyndon-Smith]

Managed to get the portech 370 up and running with asterisk (even got 
the callerid working!), but was wondering how (if) it is possible to 
send / receive sms messages through the device . All I could find 
googling was people asking how ;(

Does anyone have sms working with this device ?

Julian

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Re: [asterisk-users] Any other "free" toll free SIP providers out there?

[Tom Browning]

I tweaked the voip-info page a bit to reflect your example correctly (my
example stripped the first digit as I am using 8 as the dial prefix to toll
free via "free SIP" providers )



On Thu, Nov 20, 2008 at 11:02 AM, Atis Lezdins <[EMAIL PROTECTED]> wrote:

>
> Wow, that's helpful.
>
> I googled a bit, and found this lost page:
>
> http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers
>
> So, now it's updated with FWD and IdeaSIP, and linked from "VoIP
> Service Providers"
>
> Perhaps anyone who uses them can check examples - the ${EXTEN:1} part
> seems wrong.
>
> I wonder are there any legal issues if they were included in Asterisk
> sample config? Or perhaps they could even pay for advertising to get
> included there ;-)
>
>
> Regards,
> Atis
>
> --
> Atis Lezdins,
> VoIP Project Manager / Developer,
> IQ Labs Inc,
> [EMAIL PROTECTED]
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Cell Phone: +1 800 7300689
> Work phone: +1 800 7502835
>
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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Steve Totaro]

Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.

Also, do a ping to see latency,  200ms is pretty much my standard.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

On Thu, Nov 20, 2008 at 12:16 PM, Tim Nelson <[EMAIL PROTECTED]> wrote:
> I'm not sure about the 3 second delay, but I've seen plenty of echo issues on 
> Polycom phones when the gain has been changed on the handset. Check the 
> voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're 
> not too high.
>
> You also may want to make sure there aren't any system resource constraints 
> such as high CPU usage or memory usage... :-)
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
> - "c james" <[EMAIL PROTECTED]> wrote:
>
>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>> having a conversation.  Call quality is reported as good except for
>> an
>> echo with a 3 second delay.
>>
>> Most of my searches are saying echo happens only on the PSTN piece,
>> but
>> there isn't one here.
>>
>> Can someone point me in the right direction?
>>
>> Asterisk 1.4.21.2
>> Under 40 users
>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
>> they wanted to use!)
>>

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Daniel Hazelbaker]

On Nov 20, 2008, at 9:02 AM, Olivier wrote:


2008/11/20 Daniel Hazelbaker <[EMAIL PROTECTED]>
Any reason you want to use the MAC address?  If it is just for easy
provisioning, I just put a MAC address field in the realtime SIP table
and use a php script to take the phone's MAC address and feed it the
login information it needs.

provisioning is the first reason.
I also thought it could help to separate devices, users and other  
resources.


What I currently do to separate "devices" (fax machines, modems, etc.)  
is give them actual names.  I.e. I have "northFax" and "southFax"  
defined (so I would type 'sip show peer northFax').  In my mind, and  
particularly in my use, anything that a person dial's as an extension  
is going to have a person on the other end.  Other things can have  
names because end users won't be dialing them as extensions.  The fax  
machines are "tied" to a dedicated phone number so Asterisk dial's it  
internally.


as you obviously cannot tie MAC address to a dialing string, this  
forces you to query a database somewhere for every call ...


I'm not fully convinced of this, anyway, but when I thought about  
it, I felt frightened about loosing things I'm used to ...


Correct.  We setup a macro that uses a MySQL database to handle our  
extension dialing, we don't dial by MAC address but if you were so  
inclined, I suppose you could. As far as speed goes, we query the  
database about 4-6 times for every call.  85 users, 9 telco lines,  
Dell 2950 server, and we peak at about 0.2% cpu usage.  Again for  
simplicity, having all the "front-scene" stuff match what the end-user  
is talking about is very nice.  There is no reason you couldn't do  
some naming convention like 'user', 'device',  
'other'.  That might help in your separation and wouldn't  
be too hard to figure out.


Daniel

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

SIP wrote:

> As for the current F5 SIP load balancer, we tried it a few years back
> and it was a dismal failure. It wanted to do cookie-based SIP load
> balancing and only worked with certain SIP proxies.

I assume that is because there is no way RFC-supported way to insert a 
cookie into a SIP session that persists throughout the entire exchange 
with a client, including all in-dialog requests, subsequent sessions, etc?

The only way I know of to make a cookie stick on the UAC side is to put 
an LR parameter into the route set, but that will only last within a dialog.

So, I'm assuming certain SIP proxies had proprietary ways of getting 
around that in order to work with F5?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] SIP to IAX2 with delayed echo

[Tim Nelson]

I'm not sure about the 3 second delay, but I've seen plenty of echo issues on 
Polycom phones when the gain has been changed on the handset. Check the 
voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're 
not too high.

You also may want to make sure there aren't any system resource constraints 
such as high CPU usage or memory usage... :-)

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- "c james" <[EMAIL PROTECTED]> wrote:

> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
> having a conversation.  Call quality is reported as good except for
> an
> echo with a 3 second delay.
> 
> Most of my searches are saying echo happens only on the PSTN piece,
> but
> there isn't one here.
> 
> Can someone point me in the right direction?
> 
> Asterisk 1.4.21.2
> Under 40 users
> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
> they wanted to use!)
> 
> 
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Re: [asterisk-users] setting up callback

[Mikhail (Plus Plus)]

Nobody responded, but I was able to resolve this issue the way I wanted.
In my extensions.conf I put the following:

[callback-dialtone-auth]
exten => s,1,answer()
exten => s,n,authenticate(5678)
exten => s,n,Read(fwd_callback_to)
exten => s,n,NoOP(${fwd_callback_to})
exten => s,n,Dial(SIP/[EMAIL PROTECTED])

Now when I receive a callback, I enter password and after entering 
password I enter the phone number ending with # I wish to call and it 
gets passed to callcentric SIP and all works great.

I posted the solution just in case someone else will have similar issues.

M.

Михаил (Плюс Плюс) wrote:
> Greetings Asterisk users!
> 
> I'm trying to setup Asterisk system to act as a callback system together
> with callcentric (http://callcentric.com) but it appears that I hit common
> DTMF issue and I want to workaround this problem. Basically my current
> setup is the following:
> 
> 1) I have dedicated Asterisk server that it is linked to my callcentric
> account
> 2) I have US phone number (DID) from callcentric attached to my account
> 3) I want to make calls from my cell phone to (real US) callcentric number
> and receive a callback to my cell phone number. After receiving callback,
> I enter 4-digit password to auth myself and then I get a line via DISA
> feature of Asterisk.
> 
> I guess my setup is very common, and all is great (e.g. I'm able to
> receive a callback, enter password and then get callcentric line), except
> that callcentric does not appear to be getting DTMF tones from my cell
> phone correctly and I am unable to make a call.
> I have searched all day long today and all I was able to find is that some
> people have callback DTMF working with callcentric fine and others not. I
> tweaked my sip.conf with all possible combinations of "dtmfmode" setting,
> but still no luck.
> 
> Maybe I want something strange, but it appears that in my case Asterisk is
> able to read DTMF tones correctly while making callback and asking me to
> enter password to authenticate myself (I am able to pass authentication
> process with no problems), so what I want to do is to use that instead of
> using DISA feature of Asterisk. In other words I want something like this:
> 
> -> I call callcentric from my cell
> -> Asterisk calls me back using callcentric line
> -> I enter 4-digit password to authenticate first
> -> if authentication went through, I type a phone number I wish to call
> -> Asterisk initiates a SIP call to provided phone number through
> callcentric, and all this has to work so that I can speak and hear remote
> party on my cell phone.
> 
> I hope the above scheme is clear enough to understand.
> The problem is that I cannot understand how to implement the above -
> should this be done with "WaitExten()" feature? If so, can someone share
> examples of their setup? I would appreciate any pointers to implement the
> above.
> 
> My current GSM provider in Russia is Megafon, and I believe this has
> something to do with them that DTMF tones don't get passed correctly.
> 
> Here's my current parts of config files responsible for callback:
> 
> sip.conf:
> 
> register => 1777286:[EMAIL PROTECTED]/1862772
> ...
> [callcentric]
> type=peer
> context=from-callcentric
> host=callcentric.com
> username=1777286
> secret=XXX
> fromuser=1777286
> fromdomain=callcentric.com
> disallow=all
> allow=alaw
> dtmfmode=inband
> canreinvite=no
> ;rfc2833compensate=yes
> insecure=very
> 
> 
> 
> extensions.conf:
> 
> NOTE: 1862772 is a real phone # I have in my callcentric account
> 
> [from-callcentric]
> exten => 1862772,1,NoOp(callcentric callback to ${CALLERID(num))
> exten => 1862772,2,Wait(1)
> exten => 1862772,3,system(cp /var/spool/asterisk/skelett.call
> /var/spool/asterisk/skelett.tmp.call)
> exten => 1862772,4,system(echo 'Channel:
> SIP/+${CALLERID(num)[EMAIL PROTECTED]' >>
> /var/spool/asterisk/skelett.tmp.call)
> exten => 1862772,5,system(mv /var/spool/asterisk/skelett.tmp.call
> /var/spool/asterisk/outgoing)
> exten => 1862772,6,HangUp
> 
> [callback-dialtone-auth]
> exten => s,1,answer()
> exten => s,n,authenticate(5678)
> exten => s,n,DISA(no-password,home)
> 
> 
> 
> /var/spool/asterisk/skelett.call:
> 
> Context: callback-dialtone-auth
> Extension: s
> MaxRetries: 2
> RetryTime: 1
> 
> 
> 
> Thank you,
> Mikhail.
> 
> 
> 
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Re: [asterisk-users] two sip listening ports for single asterisk

[Matthew J. Roth]

Mike wrote:
> I tried using this iptables sample, and did not see duplicate packets
> on '--to-ports' port
>
> Has some verified this is working for them?
>
> I listened on both ports with tcpdump command.

Mike,

I can confirm that it's working.  Admittedly, I never looked at the 
packets with tcpdump because this *just worked* for me.  Calls that were 
sent to both ports (5060 and 5062) made it to Asterisk which was only 
listening on port 5060.  What's your experience with actual calls?

As the original poster, I understand if you want third-party 
verification.  I *thought* this was a slamdunk but I'm not an iptables 
guru so I'd like it, too.

What does the output of "iptables-save" and "lsmod" look like?  Here's 
mine, trimmed for relevancy:

[EMAIL PROTECTED] ~]# iptables-save
# Generated by iptables-save v1.3.5 on Thu Nov 20 12:03:21 2008
*nat
:PREROUTING ACCEPT [5579:1727747]
:POSTROUTING ACCEPT [1943:176116]
:OUTPUT ACCEPT [1943:176116]
-A PREROUTING -i eth2 -p udp -m udp --dport 5062 -j REDIRECT --to-ports 5060
COMMIT
# Completed on Thu Nov 20 12:03:21 2008

[EMAIL PROTECTED] ~]# lsmod
Module  Size  Used by
ip_conntrack_netbios_ns36033  0
ipt_REDIRECT   35009  1
xt_tcpudp  36417  1
iptable_nat40773  1
ip_nat 53101  2 ipt_REDIRECT,iptable_nat
ip_conntrack   91237  3 
ip_conntrack_netbios_ns,iptable_nat,ip_nat
nfnetlink  40457  2 ip_nat,ip_conntrack
ip_tables  55329  1 iptable_nat
x_tables   50377  4 
ipt_REDIRECT,xt_tcpudp,iptable_nat,ip_tables

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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[asterisk-users] SIP to IAX2 with delayed echo

[c james]

A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation.  Call quality is reported as good except for an
echo with a 3 second delay.

Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.

Can someone point me in the right direction?

Asterisk 1.4.21.2
Under 40 users
Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
they wanted to use!)


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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Olivier]

2008/11/20 Daniel Hazelbaker <[EMAIL PROTECTED]>

> It will auto-complete if you hit tab, just like the shell.  But I
> would recommend against it.  I can't really think of a good reason to
> do it.  'sip show peer 268' I can remember to see that status of
> extension 268 when somebody calls and says "I can't dial 268".
> Whereas 'sip show peer 00147...wtf was his MAC address again?', I have
> to lookup the extension somewhere and find the MAC address.


I mostly agree with that and that's what kept me from doing it earlier.

>
>
> Any reason you want to use the MAC address?  If it is just for easy
> provisioning, I just put a MAC address field in the realtime SIP table
> and use a php script to take the phone's MAC address and feed it the
> login information it needs.


provisioning is the first reason.
I also thought it could help to separate devices, users and other resources.

as you obviously cannot tie MAC address to a dialing string, this forces you
to query a database somewhere for every call ...

I'm not fully convinced of this, anyway, but when I thought about it, I felt
frightened about loosing things I'm used to ...

>
>
> Daniel
>
> > Hi,
> >
> > For a long time, I was wondering if I should use MAC address instead
> > of Extension number to identify SIP endpoints (as I'm mostly not
> > using softphones).
> >
> > Before diving into this, I wondered how people using MAC address are
> > using CLI as it seems more natural and simple to type
> > "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
> > Is there something obvious I'm missing (auto-completion ?
> > aliasing ? ...) ?
> >
> > Cheers
> > ___
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> >
> > 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] Macro conversion in 1.6

[Joseph L. Casale]

>AFAIR it was mentioned in UPGRADE.txt that argument separator was
>changed from pipe to comma. Unless you read it, you might also
>experience lot of other problems.

Whoops, missed that! I did see the suggestion on GoSub's but as it
stated Macros would still be supported I neglected to attempt to rewrite
it yet.

There isn't a lot of info on GoSub as its new, so I figured I would just wait.

Thanks for the pointer!
jlc

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Olivier]

2008/11/20 Philipp Kempgen <[EMAIL PROTECTED]>

> Olivier schrieb:
>
> > For a long time, I was wondering if I should use MAC address instead of
> > Extension number to identify SIP endpoints (as I'm mostly not using
> > softphones).
> >
> > Before diving into this, I wondered how people using MAC address are
> using
> > CLI as it seems more natural and simple to type
> > "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
> > Is there something obvious I'm missing (auto-completion ? aliasing ? ...)
> ?
>
> SIP accounts are about users (not extensions or devices).


I would incline to say entries in sip.conf mostly relates to devices (hence
passwords which are automatically sent when devices register) and I agree
that SIP accounts are about users ...

>
>
>
>   Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Philipp Kempgen]

Olivier schrieb:

> For a long time, I was wondering if I should use MAC address instead of
> Extension number to identify SIP endpoints (as I'm mostly not using
> softphones).
> 
> Before diving into this, I wondered how people using MAC address are using
> CLI as it seems more natural and simple to type
> "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
> Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ?

SIP accounts are about users (not extensions or devices).


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Steve Howes]

On 20 Nov 2008, at 16:14, Daniel Hazelbaker wrote:
> Any reason you want to use the MAC address?

Bet he used to use Cisco ;)

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Re: [asterisk-users] Using MAC or extension number as SIP identifier

[Daniel Hazelbaker]

It will auto-complete if you hit tab, just like the shell.  But I  
would recommend against it.  I can't really think of a good reason to  
do it.  'sip show peer 268' I can remember to see that status of  
extension 268 when somebody calls and says "I can't dial 268".   
Whereas 'sip show peer 00147...wtf was his MAC address again?', I have  
to lookup the extension somewhere and find the MAC address.

Any reason you want to use the MAC address?  If it is just for easy  
provisioning, I just put a MAC address field in the realtime SIP table  
and use a php script to take the phone's MAC address and feed it the  
login information it needs.

Daniel

> Hi,
>
> For a long time, I was wondering if I should use MAC address instead  
> of Extension number to identify SIP endpoints (as I'm mostly not  
> using softphones).
>
> Before diving into this, I wondered how people using MAC address are  
> using CLI as it seems more natural and simple to type
> "sip show peer 4566"  as opposed to "sip show peer 00147F784512".
> Is there something obvious I'm missing (auto-completion ?  
> aliasing ? ...) ?
>
> Cheers
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>   http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Macro conversion in 1.6

[Atis Lezdins]

On Thu, Nov 20, 2008 at 5:57 PM, Joseph L. Casale
<[EMAIL PROTECTED]> wrote:
> I create my sip users using a common macro in 1.4:
> [internal]
> exten => 200,1,Macro(phones|200|SIP/200)
> [macro-phones]
> exten => s,1,Dial(${ARG2}|45|Tt)
> etc...
>
> But now in 1.6 this fails:
>
>-- Executing [EMAIL PROTECTED]:1] Macro("SIP/201-0942b530", 
> "phones|200|SIP/200") in new stack
> [Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context 
> 'macro-phones|200|SIP/200' for macro 'phones|200|SIP/200'
>-- Executing [EMAIL PROTECTED]:2] Wait("SIP/201-0942b530", "1") in new 
> stack
>-- Executing [EMAIL PROTECTED]:3] Playback("SIP/201-0942b530", "invalid") 
> in new stack
>--  Playing 'invalid.gsm' (language 'en')
>
> Why does the user's extension get created (all the phones work) but I can't 
> dial to it?
>

AFAIR it was mentioned in UPGRADE.txt that argument separator was
changed from pipe to comma. Unless you read it, you might also
experience lot of other problems.

It should be Macro(phones,200,SIP/200)

However it's not recommended to use macro's, you are encouraged to
convert them to GoSub's, as they now support arguments.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Load balancing Asterisk.

[Grey Man]

> 3. Incoming calls - I admit complete ignorance. I don't know
> how OpenSIPS handles incoming calls, but for those to arrive
> at the user reliably they must arrive from the same IP address
> the user is registered to. Otherwise their broadband router's
> NAT firewall will just block the connection. How does OpenSIPS
> handle this? (does it handle this??)

That's the big question!

My company uses a custom SIP Proxy and SIP Registrar so I can't speak
for the details of SER derivatives but the theory is most likely the
same.

Our SIP Registrar records the proxy the REGISTER request arrived on
and updates the Asterisk realtime database outboundproxy field with
that value. When Asterisk needs to send an incoming call to the user
it looks up the SIP username in the realtime database and sends the
call thorugh the correct Proxy which solves the NAT issue you mention.

One trick for young players here is that the outboundproxyport setting
is broken in Asterisk so your Proxy will have to run on port 5060.

Regards,

Greyman.

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Re: [asterisk-users] Any other "free" toll free SIP providers out there?

[Atis Lezdins]

On Thu, Nov 20, 2008 at 2:50 PM, SIP <[EMAIL PROTECTED]> wrote:
> Tom Browning wrote:
>>
>> FWD (Free World Dialup) allows any SIP call to US toll free numbers
>> via [EMAIL PROTECTED] 
>> This works WITHOUT the need to be registered at FWD so in my dialplan
>> I have something like:
>>
>> exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
>> )
>> exten => _8.,2,Hangup
>>
>>
>> And I just dial 8-1-8xxyyy and presto ...  calls go through just
>> fine 99% of the time.
>>
>> I'm wondering if there are any other providers out there that allow
>> calls to toll free numbers without the need of being registered?  I'd
>> like to have a backup or two.
>>
>> Tom
>> 
>>
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>
> IdeaSIP doesn't require registration for free Toll-Free.
> [EMAIL PROTECTED]
>

Wow, that's helpful.

I googled a bit, and found this lost page:

http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers

So, now it's updated with FWD and IdeaSIP, and linked from "VoIP
Service Providers"

Perhaps anyone who uses them can check examples - the ${EXTEN:1} part
seems wrong.

I wonder are there any legal issues if they were included in Asterisk
sample config? Or perhaps they could even pay for advertising to get
included there ;-)


Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] Using MAC or extension number as SIP identifier

[Olivier]

Hi,

For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).

Before diving into this, I wondered how people using MAC address are using
CLI as it seems more natural and simple to type
"sip show peer 4566"  as opposed to "sip show peer 00147F784512".
Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ?

Cheers
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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:
> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
> 
>> I am not agreed on point 2:
>> If I understood how to install opensips + heartbeat WITHOUT
>> knowing any
>> program (opensips ? heartbear ?) or programming
>> language(hell yes!) in a
>> week ( just knew what's invite and bye ;) a more aware
>> IT professional could
>> do it in 2 days
> 
> I'm actually referring mostly to the need to build, install,
> and maintain another set (2?) of Linux boxes. The software is
> the easy part.

As someone who hates dealing with hardware, I can relate and appreciate 
why this is a pain.

But it's a lot easier than setting up the alternatives!

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Macro conversion in 1.6

[Joseph L. Casale]

I create my sip users using a common macro in 1.4:
[internal]
exten => 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten => s,1,Dial(${ARG2}|45|Tt)
etc...

But now in 1.6 this fails:

-- Executing [EMAIL PROTECTED]:1] Macro("SIP/201-0942b530", 
"phones|200|SIP/200") in new stack
[Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context 
'macro-phones|200|SIP/200' for macro 'phones|200|SIP/200'
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/201-0942b530", "1") in new stack
-- Executing [EMAIL PROTECTED]:3] Playback("SIP/201-0942b530", "invalid") 
in new stack
--  Playing 'invalid.gsm' (language 'en')

Why does the user's extension get created (all the phones work) but I can't 
dial to it?

Thanks!
jlc

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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

--- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:

> I am not agreed on point 2:
> If I understood how to install opensips + heartbeat WITHOUT
> knowing any
> program (opensips ? heartbear ?) or programming
> language(hell yes!) in a
> week ( just knew what's invite and bye ;) a more aware
> IT professional could
> do it in 2 days

I'm actually referring mostly to the need to build, install,
and maintain another set (2?) of Linux boxes. The software is
the easy part.

Granted, if that's what we need to do - that's what we'll do.

--
Nitzan Kon, CEO
Future Nine Corporation
http://www.future-nine.com

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Re: [asterisk-users] Role of asterisk

[Valentin Bud]

On Thu, Nov 20, 2008 at 5:35 PM, Bill Andersen <[EMAIL PROTECTED]> wrote:
> Jared Smith had written:
>> To answer the second portion of your question (which I forgot to do in
>> my earlier email)... yes, Asterisk can be a registration server as well.
>>
>> --
>> Jared Smith
>> Training Manager
>> Digium, Inc.
>
> Valentin Bud wrote:
>> Hello Mr. Smith,
>  
>>If you know any kind of books that are suitable for a beginner please let
> me know.
>
> Hi Valentin,

Hi Bill,

>
> I really like the book "Asterisk: The Future of Telephony, Second Edition"
> authored by some guy named Jared Smith...  Coincidence?  You be the judge...

Already read that. The book is great for a beginner and a must-have if
you are in the
VoIP-with-asterisk business. I'm looking for a SIP protocol book to
understand a problem
i have.

Speaking of which Mr. Smith, the book you told me about is very good.
I didn't had
the time to read it all but the introduction and the content i saw is
very nice. Takes you
through telephony a little bit and SIP especially.

PS: shame on me cause i didn't notice the name of Mr. Smith, thought i know it
from somewhere but couldn't make the connection.

a great day,
v

>
> http://oreilly.com/catalog/9780596510480/
>
> :)
>
>
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Re: [asterisk-users] Load balancing Asterisk.

[Grygoriy Dobrovolskyy]

2. Overkill to install and maintain (if we can get a simpler
solution)

I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT knowing any
program (opensips ? heartbear ?) or programming language(hell yes!) in a
week ( just knew what's invite and bye ;) a more aware IT professional could
do it in 2 days
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Re: [asterisk-users] Load balancing Asterisk.

[SIP]

Alex Balashov wrote:
> I was about to say, I'm sure F5 can do it... but...
>
>  > price was over 6 figures
>
> Why??!
>
> It's spending money on these types of things when they are unnecessary 
> that is the undoing of every struggling VoIP provider I watch, in the 
> misguided belief that only will half a million dollars get you 
> "enterprise strength."  That was the conventional wisdom about Linux ten 
> years ago too.  Who's saying that now?  Ditto.
>
>   
F5 has ALWAYS been overpriced.

Incidentally, anyone who wants to know, F5 is a unix-based box, just
like the others. Last we used the F5s, they were all running a slightly
modified BSDI. And only slightly modified in packaging.

As for the current F5 SIP load balancer, we tried it a few years back
and it was a dismal failure. It wanted to do cookie-based SIP load
balancing and only worked with certain SIP proxies.

N.

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:

> My concerns with OpenSIPS:
> 
> 1. It's a software based solution, which means higher chance
> of software-related failure, and higher chance of failure due
> to problems with the Linux box hosting it.

A little bit of proper engineering will overcome that reasonably.

> 2. Overkill to install and maintain (if we can get a simpler
> solution)

Really?

It is, admittedly, a somewhat recondite product, but you don't have to 
build everything you run into your core competency;  you can divest 
yourself of some parts of your infrastructure and streamline and all 
that and get someone else to do it, like a real Enterprise.  :-)

Secondly, as difficult as it may be, I can't imagine anything simpler to 
accomplish what you're looking for.  The logic required is quite granular.

> 3. Incoming calls - I admit complete ignorance. I don't know
> how OpenSIPS handles incoming calls, but for those to arrive
> at the user reliably they must arrive from the same IP address
> the user is registered to. Otherwise their broadband router's
> NAT firewall will just block the connection. How does OpenSIPS
> handle this? (does it handle this??)

What role are you envisioning the proxy to be in here?  If it's a 
registrar, it will have their IP information in the stored contact URI. 
  If not, the calls can be sent somewhere else for resolution. 
Something, somewhere must know how to contact the user, yes.

-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] jitterbuffer

[farah . auf]

 i want to get some statistics about the call quality with asterisk.
 I used the following command: iax2 show netstats
 and the result changes depending on the configuration of iax.conf.

 When i enable jitterbuffer=yes and forcejitterbuffer=yes, i get the
 following result:

 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-6207  15002   42 0   0 00  00
   0 0   0 00  0
 1 active IAX channel
 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-6207222   43 1   0 00  10
  40 0   0 00  0
 1 active IAX channel
 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-6207222   43 1   0 00  10
  40 0   0 00  0
 1 active IAX channel


 and when i disable it i get:

 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-11724 1500   -10-1  -1 0   -1  00
   0 0   0 00  0
 1 active IAX channel
 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-11724   20   -10-1  -1 0   -1  10
  40 0   0 00  0
 1 active IAX channel
 voip*CLI> iax2 show netstats
  LOCAL -
  REMOTE 
 ChannelRTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit
 Del  Lost   %  Drop  OOO  Kpkts
 IAX2/i55333-11724   20   -10-1  -1 0   -1  10
  40 0   0 00  0
 1 active IAX channel


 Should i enable it or not?
 Is there any threshold for the field "lost" and "%" for which we can say
 that the call quality is bad?

 Thank you

 F.A
>
> ==
>
> --
>  (0095170) blitzrage (administrator) - 2008-11-20 08:47
>  http://bugs.digium.com/view.php?id=13937#c95170
> --
> This is not a bug, but rather a support issue. Please use the appropriate
> venues for support such as the #asterisk IRC channel, or the asterisk-users
> mailing list. Thanks!
>
> Issue History
> Date ModifiedUsername   FieldChange
> ==
> 2008-11-20 08:47 blitzrage  Note Added: 0095170
> 2008-11-20 08:47 blitzrage  Status   new => closed
> 2008-11-20 08:47 blitzrage  Resolution   open => no change
> required
> ==
>
>



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Re: [asterisk-users] Role of asterisk

[Bill Andersen]

Jared Smith had written:
> To answer the second portion of your question (which I forgot to do in
> my earlier email)... yes, Asterisk can be a registration server as well.
>
> --
> Jared Smith
> Training Manager
> Digium, Inc.

Valentin Bud wrote:
> Hello Mr. Smith,
  
>If you know any kind of books that are suitable for a beginner please let
me know.

Hi Valentin,

I really like the book "Asterisk: The Future of Telephony, Second Edition"
authored by some guy named Jared Smith...  Coincidence?  You be the judge...

http://oreilly.com/catalog/9780596510480/

:)


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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

I was about to say, I'm sure F5 can do it... but...

 > price was over 6 figures

Why??!

It's spending money on these types of things when they are unnecessary 
that is the undoing of every struggling VoIP provider I watch, in the 
misguided belief that only will half a million dollars get you 
"enterprise strength."  That was the conventional wisdom about Linux ten 
years ago too.  Who's saying that now?  Ditto.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

N,

SIP-aware LBs do exist - but way way out of my price range.

Alex, 

Remember we are an Asterisk-based provider. I'm not going
to drop enough money on a load balancer to go bankrupt. ;) That's
exactly why I'm wondering if it's possible to do this with a
DUMB load balancer. i.e. one that would cost about the same as
building another Linux box for OpenSIPS.

I don't need a million concurrent connections. I'd be perfectly
happy with a fraction of that. Not looking to replace AT&T here,
just looking for something simple that will work reliably. :)

My concerns with OpenSIPS:

1. It's a software based solution, which means higher chance
of software-related failure, and higher chance of failure due
to problems with the Linux box hosting it.
2. Overkill to install and maintain (if we can get a simpler
solution)
3. Incoming calls - I admit complete ignorance. I don't know
how OpenSIPS handles incoming calls, but for those to arrive
at the user reliably they must arrive from the same IP address
the user is registered to. Otherwise their broadband router's
NAT firewall will just block the connection. How does OpenSIPS
handle this? (does it handle this??)

Thanks!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

--- On Thu, 11/20/08, SIP <[EMAIL PROTECTED]> wrote:

> Unless the LB is SIP-aware, and can maintain a SIP session,
> I don't see
> how it would work. As the SIP command stream sends discrete
> commands,
> without some sort of basic level of session awareness,
> there's no
> guarantee over a reasonable-length call that the INVITE and
> BYE would
> even get sent to the same Asterisk box. If there are
> on-hold messages or
> transfers occurring, you add even more possibility of error
> into the
> mix.  Now... you could do some sort of VERY long session
> timeout, but
> overall, that's a hack that's going to drop your
> concurrent connection
> count faster than using a smaller box would.
> 
> I don't know of any functioning, SIP-aware load
> balancers at the moment.
> Doesn't mean they don't exist. I just can't
> think of any off the top of
> my head.
> 
> N.
> 
> 
> 
> Nitzan Kon wrote:
> > Alex,
> >
> > I realize and agree that "hardware" load
> balancers are actually
> > software based. I'm less concerned about that and
> more about the
> > general specs:
> >
> > Foundry ServerIron XL: rated for 1,000,000 concurrent
> connections
> > Linux box where OpenSIPS is sitting: rated for ...???
> >
> > Not to mention a simple rule on a load balancer would
> be much,
> > much easier to implement. All I need is IP-based load
> balancing
> > so installing and maintaining OpenSIPS is an overkill.
> >
> > Again, I appreciate the feedback but I am not asking
> nor looking
> > for a software solution. My question is simple:
> >
> > Will a HARDWARE load balancer work? any reason why it
> WON'T work?
> >
> > Thanks!
> >
> >
> > --- On Thu, 11/20/08, Alex Balashov
> <[EMAIL PROTECTED]> wrote:
> >
> >   
> >> What do you mean by "hardware" options? 
> There are
> >> no ASIC-assisted SIP load balancers out there. 
> :-)  The
> >> embedded "hardware-based" options are
> load
> >> balancers built just like PCs - often on top of a
> UNIX
> >> kernel - that run a software application-aware
> load
> >> balancing suite.
> >>
> >> Your best bet is a proxy for the round-robin part,
> and
> >> Linux-HA for the high availability of the proxy,
> as Grygoriy
> >> suggested.
> >>
> >> Nitzan Kon wrote:
> >>
> >> 
> >>> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
> >>>   
> >> <[EMAIL PROTECTED]> wrote:
> >> 
>  2 openser servers with 3 ip adresses (1
> virtual) +
>  heartbeat to ensure the
>  failover + watchdog to ensure if
>  
> >> opensips/kamalio/openser
> >> 
>  crashes a nice
>  failover & reboot, it is working
> stable here
>  (dispatching to 10 servers +
>  owners DID dispatch to their respective
> servers)
> 
>  join #opensips on freenode if you need
> more info.
>  
> >>> Thanks for the info. :)
> >>>
> >>> I want to stay away from software solutions
> however.
> >>>   
> >> Are there
> >> 
> >>> any hardware solutions? would a plain load
> balancer
> >>>   
> >> work?
> >> 
> >>> If we can't get it working with a LB
> we'll
> >>>   
> >> look at OpenSIPS,
> >> 
> >>> but I'd like to explore hardware options
> first.
> >>>
> >>> Thanks!
> >>>
> >>> --
> >>> Nitzan Kon, CEO
> >>> Future Nine Corporation
> >>> www.future-nine.com
> >>>   
> >
> >
> >
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> >   
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Re: [asterisk-users] Load balancing Asterisk.

[Grey Man]

This baby talks about being able to do hardware SIP load balancing.

http://www.f5.com/news-press-events/press/2007/20070212.html

I've never used an f5 product so I can't provide any comments from
experience. I did look at an f5 load balancer product once and the
price was over 6 figures that was a few years ago though.

Regards,

Greyman.

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

The solution to make this work and still work "statelessly" is to hash 
various unique identifying bits of the SIP headers without maintaining 
transactional, session or dialog information as such.

SIP wrote:

> Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
> how it would work. As the SIP command stream sends discrete commands,
> without some sort of basic level of session awareness, there's no
> guarantee over a reasonable-length call that the INVITE and BYE would
> even get sent to the same Asterisk box. If there are on-hold messages or
> transfers occurring, you add even more possibility of error into the
> mix.  Now... you could do some sort of VERY long session timeout, but
> overall, that's a hack that's going to drop your concurrent connection
> count faster than using a smaller box would.
> 
> I don't know of any functioning, SIP-aware load balancers at the moment.
> Doesn't mean they don't exist. I just can't think of any off the top of
> my head.
> 
> N.
> 
> 
> 
> Nitzan Kon wrote:
>> Alex,
>>
>> I realize and agree that "hardware" load balancers are actually
>> software based. I'm less concerned about that and more about the
>> general specs:
>>
>> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
>> Linux box where OpenSIPS is sitting: rated for ...???
>>
>> Not to mention a simple rule on a load balancer would be much,
>> much easier to implement. All I need is IP-based load balancing
>> so installing and maintaining OpenSIPS is an overkill.
>>
>> Again, I appreciate the feedback but I am not asking nor looking
>> for a software solution. My question is simple:
>>
>> Will a HARDWARE load balancer work? any reason why it WON'T work?
>>
>> Thanks!
>>
>>
>> --- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
>>
>>   
>>> What do you mean by "hardware" options?  There are
>>> no ASIC-assisted SIP load balancers out there.  :-)  The
>>> embedded "hardware-based" options are load
>>> balancers built just like PCs - often on top of a UNIX
>>> kernel - that run a software application-aware load
>>> balancing suite.
>>>
>>> Your best bet is a proxy for the round-robin part, and
>>> Linux-HA for the high availability of the proxy, as Grygoriy
>>> suggested.
>>>
>>> Nitzan Kon wrote:
>>>
>>> 
 --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
   
>>> <[EMAIL PROTECTED]> wrote:
>>> 
> 2 openser servers with 3 ip adresses (1 virtual) +
> heartbeat to ensure the
> failover + watchdog to ensure if
> 
>>> opensips/kamalio/openser
>>> 
> crashes a nice
> failover & reboot, it is working stable here
> (dispatching to 10 servers +
> owners DID dispatch to their respective servers)
>
> join #opensips on freenode if you need more info.
> 
 Thanks for the info. :)

 I want to stay away from software solutions however.
   
>>> Are there
>>> 
 any hardware solutions? would a plain load balancer
   
>>> work?
>>> 
 If we can't get it working with a LB we'll
   
>>> look at OpenSIPS,
>>> 
 but I'd like to explore hardware options first.

 Thanks!

 --
 Nitzan Kon, CEO
 Future Nine Corporation
 www.future-nine.com
   
>>
>>
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Load balancing Asterisk.

[SIP]

Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would work. As the SIP command stream sends discrete commands,
without some sort of basic level of session awareness, there's no
guarantee over a reasonable-length call that the INVITE and BYE would
even get sent to the same Asterisk box. If there are on-hold messages or
transfers occurring, you add even more possibility of error into the
mix.  Now... you could do some sort of VERY long session timeout, but
overall, that's a hack that's going to drop your concurrent connection
count faster than using a smaller box would.

I don't know of any functioning, SIP-aware load balancers at the moment.
Doesn't mean they don't exist. I just can't think of any off the top of
my head.

N.



Nitzan Kon wrote:
> Alex,
>
> I realize and agree that "hardware" load balancers are actually
> software based. I'm less concerned about that and more about the
> general specs:
>
> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
> Linux box where OpenSIPS is sitting: rated for ...???
>
> Not to mention a simple rule on a load balancer would be much,
> much easier to implement. All I need is IP-based load balancing
> so installing and maintaining OpenSIPS is an overkill.
>
> Again, I appreciate the feedback but I am not asking nor looking
> for a software solution. My question is simple:
>
> Will a HARDWARE load balancer work? any reason why it WON'T work?
>
> Thanks!
>
>
> --- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
>   
>> What do you mean by "hardware" options?  There are
>> no ASIC-assisted SIP load balancers out there.  :-)  The
>> embedded "hardware-based" options are load
>> balancers built just like PCs - often on top of a UNIX
>> kernel - that run a software application-aware load
>> balancing suite.
>>
>> Your best bet is a proxy for the round-robin part, and
>> Linux-HA for the high availability of the proxy, as Grygoriy
>> suggested.
>>
>> Nitzan Kon wrote:
>>
>> 
>>> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
>>>   
>> <[EMAIL PROTECTED]> wrote:
>> 
 2 openser servers with 3 ip adresses (1 virtual) +
 heartbeat to ensure the
 failover + watchdog to ensure if
 
>> opensips/kamalio/openser
>> 
 crashes a nice
 failover & reboot, it is working stable here
 (dispatching to 10 servers +
 owners DID dispatch to their respective servers)

 join #opensips on freenode if you need more info.
 
>>> Thanks for the info. :)
>>>
>>> I want to stay away from software solutions however.
>>>   
>> Are there
>> 
>>> any hardware solutions? would a plain load balancer
>>>   
>> work?
>> 
>>> If we can't get it working with a LB we'll
>>>   
>> look at OpenSIPS,
>> 
>>> but I'd like to explore hardware options first.
>>>
>>> Thanks!
>>>
>>> --
>>> Nitzan Kon, CEO
>>> Future Nine Corporation
>>> www.future-nine.com
>>>   
>
>
>
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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:

> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
> Linux box where OpenSIPS is sitting: rated for ...???

Because OpenSER's load balancer is hash-based and not stateful, it is 
rated for far, far more than that.

-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

Alex,

I realize and agree that "hardware" load balancers are actually
software based. I'm less concerned about that and more about the
general specs:

Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???

Not to mention a simple rule on a load balancer would be much,
much easier to implement. All I need is IP-based load balancing
so installing and maintaining OpenSIPS is an overkill.

Again, I appreciate the feedback but I am not asking nor looking
for a software solution. My question is simple:

Will a HARDWARE load balancer work? any reason why it WON'T work?

Thanks!


--- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:

> What do you mean by "hardware" options?  There are
> no ASIC-assisted SIP load balancers out there.  :-)  The
> embedded "hardware-based" options are load
> balancers built just like PCs - often on top of a UNIX
> kernel - that run a software application-aware load
> balancing suite.
> 
> Your best bet is a proxy for the round-robin part, and
> Linux-HA for the high availability of the proxy, as Grygoriy
> suggested.
> 
> Nitzan Kon wrote:
> 
> > --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
> <[EMAIL PROTECTED]> wrote:
> > 
> >> 2 openser servers with 3 ip adresses (1 virtual) +
> >> heartbeat to ensure the
> >> failover + watchdog to ensure if
> opensips/kamalio/openser
> >> crashes a nice
> >> failover & reboot, it is working stable here
> >> (dispatching to 10 servers +
> >> owners DID dispatch to their respective servers)
> >> 
> >> join #opensips on freenode if you need more info.
> > 
> > Thanks for the info. :)
> > 
> > I want to stay away from software solutions however.
> Are there
> > any hardware solutions? would a plain load balancer
> work?
> > 
> > If we can't get it working with a LB we'll
> look at OpenSIPS,
> > but I'd like to explore hardware options first.
> > 
> > Thanks!
> > 
> > --
> > Nitzan Kon, CEO
> > Future Nine Corporation
> > www.future-nine.com



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Re: [asterisk-users] puzzle

[Steve Totaro]

Always a self appoited list Nazi.  If it bothers you, then don't
bother reading.

On Thu, Nov 20, 2008 at 7:23 AM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> Is this a competition about how many levels of quotes the list
> can handle or something? SCNR.  ;-)
>
>
> Steve Totaro schrieb:
>> On Wed, Nov 19, 2008 at 8:58 PM, Jeff LaCoursiere <[EMAIL PROTECTED]> wrote:
>>> On Wed, 19 Nov 2008, Steve Totaro wrote:
 On Wed, Nov 19, 2008 at 8:29 PM, Jeff LaCoursiere <[EMAIL PROTECTED]> 
 wrote:
> On Wed, 19 Nov 2008, Steve Totaro wrote:
>> On Wed, Nov 19, 2008 at 8:02 PM, Jeff LaCoursiere <[EMAIL PROTECTED]> 
>> wrote:
>>> On Wed, 19 Nov 2008, Steve Totaro wrote:
 On Wed, Nov 19, 2008 at 7:19 PM, Jeff LaCoursiere <[EMAIL PROTECTED]> 
 wrote:
> On Wed, 19 Nov 2008, Steve Totaro wrote:
>> On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere <[EMAIL 
>> PROTECTED]> wrote:
>>> On Wed, 19 Nov 2008, Danny Nicholas wrote:
>
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jeff
 LaCoursiere
 On Wed, 19 Nov 2008, Danny Nicholas wrote:
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jeff
> LaCoursiere
>
> On Wed, 19 Nov 2008, Danny Nicholas wrote:
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Jeff
>> LaCoursiere
>
>> ___
>> ___
> ___
> ___
 ___
 ___
>>> ___
>> ___
> ___
 ___
>>> ___
>> ___
> ___
 ___
>>> ___
>
>
>   Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
> ___
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] Load balancing Asterisk.

[Jeff LaCoursiere]

Hardware solutions are of course simply packaged software solutions. 
Personally I would go with something that has this wonderful support base 
and quick solutions versus dealing with a vendor.  You did mention that 
price was a consideration, right?

j

On Thu, 20 Nov 2008, Nitzan Kon wrote:

> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
>
>> 2 openser servers with 3 ip adresses (1 virtual) +
>> heartbeat to ensure the
>> failover + watchdog to ensure if opensips/kamalio/openser
>> crashes a nice
>> failover & reboot, it is working stable here
>> (dispatching to 10 servers +
>> owners DID dispatch to their respective servers)
>>
>> join #opensips on freenode if you need more info.
>
> Thanks for the info. :)
>
> I want to stay away from software solutions however. Are there
> any hardware solutions? would a plain load balancer work?
>
> If we can't get it working with a LB we'll look at OpenSIPS,
> but I'd like to explore hardware options first.
>
> Thanks!
>
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
>
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Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

[Doug Lytle]

Simith Nambiar wrote:
> Hello Darrin / Doug,
>  Thank you for your response, i find 
> that the Read Aplication blocks for input and  returns when a DTMF is 
> dialled, which is fine.
> My problem is that when i use the Dial Application , it is blocking too, 
> so wheee do i put the Read call in my extensions.conf, this is how it looks.
>
> exten => 807,1,Dial(SIP/807)
> exten => 807,n,Hangup()
>
> Where can i put the below Read ? 
> exten => 807 ,n, Read(DIGITLIST,,1)
>   

;**
;* Get number from user
;**

exten => 807,1,Answer()
exten => 807,n,Read(get-room-num|conf-getconfno)

;***
;*  Echo that number back to the console
;***

exten => 807,n,NoOP(${conf-getchannel})

;
;* Dial extension 807
;

exten => 807,n,Dial(SIP/807)

;***
;* Play back entered info to 807
;***

exten => 807,n,Playback(${conf-getchannel})

;
;* Hangup
;

exten => 807,n,Hangup()

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] Voicemail in Real Time

[Philipp Kempgen]

Ali Jawad schrieb:

> I do have asterisk running in real time I do want to add voicemail to real
> time. I did follow :
> 
> http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail
> 
> However when I do try to make a voicemail I do get :
> 
> [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible
> codecs, not accepting this offer!
> -- Executing [EMAIL PROTECTED]:1]
> VoiceMail("SIP/alijawad-08aaf0f0", "[EMAIL PROTECTED]|u") in new stack
> [Nov 20 12:17:08] WARNING[22277]: app_voicemail.c:2862 leave_voicemail: No
> entry in voicemail config file for '999alijawad'
> -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/alijawad-08aaf0f0",
> "") in new stack
>   == Spawn extension (a2billing, 999alijawad, 2) exited non-zero on
> 'SIP/alijawad-08aaf0f0'
> 
> Even though I do have 999alijawad as a mailbox in voicemail_users
> 
> 
> If I do setup the mailbox in voicemail.conf it works fine.
> [a2billing]
> 999alijawad => 123456, alijawad, [EMAIL PROTECTED]
> 
> I did setup extconfig.conf as it should be:
> voicemail => mysql,mya2billing,voicemail_users

Do you have something like

[default]
switch => Realtime

in voicemail.conf?

What's the output of
voicemail show users for default
on the Asterisk CLI?


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

What do you mean by "hardware" options?  There are no ASIC-assisted SIP 
load balancers out there.  :-)  The embedded "hardware-based" options 
are load balancers built just like PCs - often on top of a UNIX kernel - 
that run a software application-aware load balancing suite.

Your best bet is a proxy for the round-robin part, and Linux-HA for the 
high availability of the proxy, as Grygoriy suggested.

Nitzan Kon wrote:

> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
> 
>> 2 openser servers with 3 ip adresses (1 virtual) +
>> heartbeat to ensure the
>> failover + watchdog to ensure if opensips/kamalio/openser
>> crashes a nice
>> failover & reboot, it is working stable here
>> (dispatching to 10 servers +
>> owners DID dispatch to their respective servers)
>>
>> join #opensips on freenode if you need more info.
> 
> Thanks for the info. :)
> 
> I want to stay away from software solutions however. Are there
> any hardware solutions? would a plain load balancer work?
> 
> If we can't get it working with a LB we'll look at OpenSIPS,
> but I'd like to explore hardware options first.
> 
> Thanks!
> 
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

--- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:

> 2 openser servers with 3 ip adresses (1 virtual) +
> heartbeat to ensure the
> failover + watchdog to ensure if opensips/kamalio/openser
> crashes a nice
> failover & reboot, it is working stable here
> (dispatching to 10 servers +
> owners DID dispatch to their respective servers)
> 
> join #opensips on freenode if you need more info.

Thanks for the info. :)

I want to stay away from software solutions however. Are there
any hardware solutions? would a plain load balancer work?

If we can't get it working with a LB we'll look at OpenSIPS,
but I'd like to explore hardware options first.

Thanks!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

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Re: [asterisk-users] Load balancing Asterisk.

[Grygoriy Dobrovolskyy]

2008/11/20 Nitzan Kon <[EMAIL PROTECTED]>

> Hello!
>
> We're looking for a solution to reliably load balance our
> Asterisk boxes. So far we've been using a hodge-podge of
> directing different services to different boxes/IPs, but
> eventually I'd like to consolidate things so we can present
> a single IP address to the outside world.
>
> My question is - how do we go about doing that? I've read
> a lot of things like load-balancing via DUNDi or OpenSER,
> but it seems to me like these approaches just add to the
> list of possible failures. In other words I'd like to avoid
> software solutions.
>
> Is it possible to just put Asterisk behind a load balancer?
> I imagine most of them are optimized for web traffic rather
> than UDP voice packets. Does that matter?
>
> Would any load balancer do - or only specific models will
> work? my guess is any model will work, but some of them may
> not be able to handle the load.
>
> Any recommended models?
>
> I know there are some fancy LBs out there that can actually
> load balance based on the SIP session rather than something
> like IP, but I'm afraid to even look at the price tag. I'm
> more than fine with balancing by user IP address instead -
> if that works. :)
>
> Would appreciate any comments or ideas.
>
> Thanks!
>
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the
failover + watchdog to ensure if opensips/kamalio/openser crashes a nice
failover & reboot, it is working stable here (dispatching to 10 servers +
owners DID dispatch to their respective servers)

join #opensips on freenode if you need more info.
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[asterisk-users] Voicemail in Real Time

[Ali Jawad]

Hi
I do have asterisk running in real time I do want to add voicemail to real
time. I did follow :

http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail

However when I do try to make a voicemail I do get :

[Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible
codecs, not accepting this offer!
-- Executing [EMAIL PROTECTED]:1]
VoiceMail("SIP/alijawad-08aaf0f0", "[EMAIL PROTECTED]|u") in new stack
[Nov 20 12:17:08] WARNING[22277]: app_voicemail.c:2862 leave_voicemail: No
entry in voicemail config file for '999alijawad'
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/alijawad-08aaf0f0",
"") in new stack
  == Spawn extension (a2billing, 999alijawad, 2) exited non-zero on
'SIP/alijawad-08aaf0f0'

Even though I do have 999alijawad as a mailbox in voicemail_users


If I do setup the mailbox in voicemail.conf it works fine.
[a2billing]
999alijawad => 123456, alijawad, [EMAIL PROTECTED]

I did setup extconfig.conf as it should be:
voicemail => mysql,mya2billing,voicemail_users

Please advice
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[asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

Hello!

We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP address to the outside world.

My question is - how do we go about doing that? I've read
a lot of things like load-balancing via DUNDi or OpenSER,
but it seems to me like these approaches just add to the
list of possible failures. In other words I'd like to avoid
software solutions.

Is it possible to just put Asterisk behind a load balancer?
I imagine most of them are optimized for web traffic rather
than UDP voice packets. Does that matter?

Would any load balancer do - or only specific models will
work? my guess is any model will work, but some of them may
not be able to handle the load.

Any recommended models?

I know there are some fancy LBs out there that can actually
load balance based on the SIP session rather than something
like IP, but I'm afraid to even look at the price tag. I'm
more than fine with balancing by user IP address instead -
if that works. :)

Would appreciate any comments or ideas.

Thanks!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

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Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

[Simith Nambiar]

Hello Darrin / Doug,
 Thank you for your response, i find 
that the Read Aplication blocks for input and  returns when a DTMF is 
dialled, which is fine.
My problem is that when i use the Dial Application , it is blocking too, 
so wheee do i put the Read call in my extensions.conf, this is how it looks.

exten => 807,1,Dial(SIP/807)
exten => 807,n,Hangup()

Where can i put the below Read ? 
exten => 807 ,n, Read(DIGITLIST,,1)

Moreover, i want to  read the DTMF from the Callee , Any help ?

Thank you,

Cheers,
Simith




Darrin Henshaw wrote:
> Yeah what Doug said ;), for more info check out: 
> http://www.voip-info.org/wiki-Asterisk+cmd+Read
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
> Sent: Thursday, November 20, 2008 8:49 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Collect digits from the Callee after the Call 
> is connected.
>
> Simith Nambiar wrote:
>   
>> Hello All,
>>   I want to collect the Digits input by the Callee after
>> the Call is connected, i use the Dial Application to connect the Caller
>>
>> 
>
>
> You'll want to look at the read application.
>
> 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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>
> This email and its attachments may be confidential and are intended solely 
> for the use of the individual or parties' to whom it is addressed. All 
> comments are solely those of the author and do not necessarily represent 
> those of Ignition.  If you are not the intended recipient of this email and 
> its attachments, you must take no action based upon them, nor must you copy 
> or show them to anyone.  Please contact the sender if you believe you have 
> received this email in error.  Thanks for considering the environmental 
> impact before printing this email.
>
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>
>   

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Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

[Darrin Henshaw]

Yeah what Doug said ;), for more info check out: 
http://www.voip-info.org/wiki-Asterisk+cmd+Read

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, November 20, 2008 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Collect digits from the Callee after the Call is 
connected.

Simith Nambiar wrote:
> Hello All,
>   I want to collect the Digits input by the Callee after
> the Call is connected, i use the Dial Application to connect the Caller
>


You'll want to look at the read application.

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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This email and its attachments may be confidential and are intended solely for 
the use of the individual or parties' to whom it is addressed. All comments are 
solely those of the author and do not necessarily represent those of Ignition.  
If you are not the intended recipient of this email and its attachments, you 
must take no action based upon them, nor must you copy or show them to anyone.  
Please contact the sender if you believe you have received this email in error. 
 Thanks for considering the environmental impact before printing this email.

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Re: [asterisk-users] Any other "free" toll free SIP providers out there?

[SIP]

Tom Browning wrote:
>
> FWD (Free World Dialup) allows any SIP call to US toll free numbers
> via [EMAIL PROTECTED]   
> This works WITHOUT the need to be registered at FWD so in my dialplan
> I have something like:
>
> exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
> )
> exten => _8.,2,Hangup
>
>
> And I just dial 8-1-8xxyyy and presto ...  calls go through just
> fine 99% of the time.
>
> I'm wondering if there are any other providers out there that allow
> calls to toll free numbers without the need of being registered?  I'd
> like to have a backup or two.
>
> Tom
> 
>
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IdeaSIP doesn't require registration for free Toll-Free. 
[EMAIL PROTECTED]

N.

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Re: [asterisk-users] Collect digits from the Callee after the Call is connected.

[Doug Lytle]

Simith Nambiar wrote:
> Hello All,
>   I want to collect the Digits input by the Callee after 
> the Call is connected, i use the Dial Application to connect the Caller 
>   


You'll want to look at the read application.

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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