Re: [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!

[Olle E Johansson]

I have many remote users, and to make the life easy I use their  
existing e.164 phone number. That way nobody of my users need to  
think what number the other party has on our system or PSTN,    
As more users you get, as less you will remember their "name".  
Therefore I tried to use the field username for this help.
It is easy to remember 49 is the (only) guy in Germany and 8621 is  
our Shanghai person, but most of the people are in 8862 (Taipei)


Is there another solution ?


using a standard configuration option for something else than it is  
intended is a dangerous path to take.


There is a setting called "setvar" that lets you set any custom  
setting you want for all incoming calls belonging to a peer or user.  
You can use this to set customer IDs, nicknames, favourite brand of  
beer - anything.


/Olle
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Re: [Asterisk-Users] PRI Presentation Restricted bit honored?

[Olle E Johansson]

Ideally I'd still like to see the number in the CDR but we can't let
users hear restricted numbers in their voicemail messages, etc.



For voicemail, I believe you pinpointed a problem. I don't think the  
voicemail application
supports the caller ID presentation setting that we do support  
elsewhere. That is propably a bug.
Please test this and open a bug report if voicemail actually e-mails  
out caller IDs that should be kept secret.


/O
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SV: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time

[jan.sarin]

>From what I understand it means that the *hardware* in your computer 
>*acknowledges* the call as soon as it is recieved and then sends it to 
>asterisk dialplan for processing.

You would essentially need to put the delay before the call ever reaches 
asterisk. So this problem isn't asterisk related... if I've understood your 
question and the answer I found correctly.

Regards,
Jan

-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Brian J. Murrell
Skickat: den 2 februari 2006 22:37
Till: asterisk-users@lists.digium.com
Ämne: Re: SV: [Asterisk-Users] delaying "answer" for a number of ringsor 
anamount of time

On Thu, 2006-02-02 at 22:08 +0100, [EMAIL PROTECTED] wrote:
> http://lists.digium.com/pipermail/asterisk-users/2005-September/125146
> .html

OK.  The hardware is a wildcard though.  How does that answer apply?
Isn't it asterisk itself that is picking that call up?  Can't it delay the pick 
up?  Maybe I am just misunderstanding your reference.

b.

--
My other computer is your Microsoft Windows server.

Brian J. Murrell
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Re: [Asterisk-Users] Asterisk video conference

[Olle E Johansson]

Asterisk's support for video over SIP is very rudimentary. Only to  
video codecs H.261, H.263, and H.263+ are supported, and even then,  
not very well. There is no support for dynamic negotiation of frame  
rates, etc. Queries to the -dev list, as to progress on these  
features were recently met with silence. We will be looking to jump  
into the project to support our own initiatives in the area of  
video in a few weeks.




You are very welcome to join the group that works with this. There is  
an open bug report in the SIP section of the bug tracker to enhance  
the videosupport setting, please test that code and add your test  
results to the bug report.


There are others out there that work with video in Asterisk and I  
have some new code from a developer that will be added to a separate  
branch soon, also for testing. Your input and additions is very much  
needed.


If we work fast, we might get some of the smaller additions into  
Asterisk 1.4, more substantial changes into the release after that.  
But that requires testing and a lot of help!


/Olle
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Re: [Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?

[Florian Overkamp]

Hi Ronald,

Ronald Wiplinger wrote:

voipbuster/   194.221.62.201  5060 UNREACHABLE
voipstunt/x 194.120.0.200   5060 



a reload shows than:

voipbuster/   80.239.235.200 5060 UNREACHABLE
voipstunt/x   194.120.0.200   5060 UNREACHABLE


Seems like voipbuster is doing round-robin DNS for redundancy. Bad 
choice with asterisk, since asterisk only looks up DNS on startup or 
reloads.


You could read out all the entries in the DNS zone and create your own 
list of entries in /etc/hosts, and then create multiple asterisk peers: 
voipbuster1, voipbuster2, etc... Then you can use regular dialplan logic 
to cycle through all of them.


Florian
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[Asterisk-Users] TDM 400 FXO FXS Test

[Jaco Maritz]

Hi

Is there a test that you can do to test your modules FXS FXO to see if they 
are blown
I have a TDM400 with 2 FXS (green) modules on the the first two ports and 2 
FXO (red) modules
on the last two ports, but I can't seem to assign a channel to them. Can any 
one help me??


Thanx
Jaco 



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Re: [Asterisk-Users] Dundi key Problem

[bbench]

On Wednesday 01 February 2006 19:48, Jonathan k. Creasy wrote:
> I am getting the following message when trying to lookup up a number via
> Dundi:
>
> Feb  1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key
> 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for
> '00:a0:c9:55:91:89'!
>
> I have created keys on each box with "astgenkey -n
> office.pbx.bluegrass.net" using the host name for each box of course.
>
> I then copied the .pub files to the /var/lib/asterisk/keys folder from
> each box to the other box.
>
> What am I missing?
inkey=office.pbx.bluegrass.net
benchev
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Re: [Asterisk-Users] Default value for ASTERISK_VERSION_NUM

[Leo Ann Boon]

Kevin P. Fleming wrote:


Leo Ann Boon wrote:


/*
* version.h
* Automatically generated
*/
#define ASTERISK_VERSION "1.2.4"
#define ASTERISK_VERSION_NUM 00



This was a bug in the Makefile; it has been corrected in Subversion 
and will part of the 1.2.5 release. Sorry for the inconvenience.


Thanks for the prompt response.

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RE : [Asterisk-Users] OT O'Reilly Asterisk TFOT

[f6hqz-m]

Hi Dave and the list,

I was at this exhibition near all the first day.
Next time, we must organise a meeting for handshaking and discussions for
Asterisk lovers during a next exhibition ?
Just by saying hello and what's happening to the list ?
It could be cool to meet us in real world  ;-)

Have you seen that 3 Asterisk servers were running during this show ?

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Dave Cotton
Envoyé : jeudi 2 février 2006 09:40
À : Asterisk List
Objet : [Asterisk-Users] OT O'Reilly Asterisk TFOT


I went to the Linux Solutions exhibition in Paris yesterday, visited the
well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours
later there was only one left. It must say something, also it was the
English version. 
-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Directed Call Pickup

[Garth van Sittert]

Show Features produces:
   Builtin Feature   Default Current
   ---   --- ---
   Pickup*8  *8
   Blind Transfer#   #
   Attended Transfer *2
   One Touch Monitor
   Disconnect Call   *   *

   Dynamic Feature   Default Current
   ---   --- ---
   (none)

   Call parking
   
   Parking extension   :   700
   Parking context :   parkedcalls
   Parked call extensions: 701-720



in extension.conf I have:
   exten => _8.,1,Pickup(${EXTEN:1})



When I dial 812, in the CLI I can see:
   Executing Pickup("SIP/29-707f", "12") in new stack


Any thoughts?

Kind Regards
Garth






Bob Goddard wrote:

On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote:
  

Hi All

I am having problems with Directed Call Pickup in Asterisk 1.2.1

If extension 100 is ringing, a user at another extension is supposed to
be able to dial *8100 and pickup the call to 100.  It isn't working for
me and I cannot figure out why.

I have in features.conf:

pickupexten = *8



At the CLI, "show features" should tell you if it is configured.
If so, you need to tell us what happens on the console.
If not, then you are liable to get asked "my car does not work,
does anyone know why?".


B

  


--
Garth van Sittert
BSc (Physics & Computer Science)
-
Mobile: +27 (0)83 791 6662
Email:  [EMAIL PROTECTED]
Phone:  08600 BITCO
Web:www.bitco.co.za 


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[Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API

[Somesh S Shanbhag]

Hi All,  Please help in this regard.  Regds, Somesh S. ShanbhagSomesh S Shanbhag <[EMAIL PROTECTED]> wrote: Hi All,  I want to do a three-party conferencing using manager api.  But I found out from the asterisk-users list that I *MUST* use  the meeting room concept.  I wanted to know wheather meeting room can be configured dynamically?  on the fly? Otherwise, configuring meeting room statically is not scalable.  Thanks Regards, Somesh S. ShanbhagBring words and photos together (easily) with  PhotoMail  - it's free and works with your Yahoo! Mail.
	
	
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Re: [Asterisk-Users] meetme and dtmf

[Francesco Peeters (Asterisk)]

On Fri, February 3, 2006 0:44, Imran Ahmed said:
>> >> Step 3 The Iax client heve to send some other DTMF to the IVR.
>> >
>> >
>> > How is the IVR still involved if the call has been transferred into a
>> > conference room?
>> >
>> The IVR records the conversation between the other partecipant to the
>> conference and wait '#' to stop recording and a '1'  to save the file.
>
> may or may not work, try at your own risk:
>
> 1) Use a sip soft phone and set the dtmf mode = inband.
> 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
> info. (this is done so that asterisk ignores the inband dtmf on the
> sip channel).
> 3) Design your dialplan such that asterisk should not depend on dtmf
> from the sip call.
> ex:
>
> exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference
> room
> exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference
> room.
> exten xxx, 3, meetme(conference room)
>
> once the sip call is in the conference then the ivr will detect dtmf
> from the audio data. Note that before the sip call is in a conference
> dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this
> is not tested and only a test can confirm if it works.
>
> drawbacks: dtmf will not be available to ivr until your call is in
> conference. asterisk will never see any dtmf (which should be okay in
> this specific case).
> dtmf tones are not squelched so the other user in the conference will
> hear dtmf tones.
>
> Imran

What I find strange is that the meetme IVR participant *does* hear DTMF
from the ZAP channel, but not from the IAX2 channel... There shouldn't be
a per channel difference in how dtmf is handled in meetme, right?...

Do you know whether the IAX2 dtmf is intercepted by meetme and handled
internally? If so you might be able to workaround by using SendDTMF() in
your meetme dialplan...

Good luck!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
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  2 Sweex HFC-PCI cards
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[Asterisk-Users] How can a caller go back to the main menu from a queue?

[Zach A]

Hi everyone,

How can I make a caller go back to the main menu if he gets tired of
waiting in a queue for too long?

Thanks,

Zeeshan A Zakaria

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RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion

[asterisk]

On Fri, 3 Feb 2006, Wai Wu wrote:
I don't think they are doing it with one Asterisk box. They did say 
"one rack of servers". Well, that might mean up to 50 computers if they 
are using blade servers.



At 11:21 AM -0800 2/2/06, William Boehlke wrote:

Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony

   

Server 5000.

   ^^^

If you look at the datasheet http://www.signate.com/pdf/TelephonyServer.pdf its
pretty clear its a cluster.

I dont think anyone would be able to route 5,000 RTP streams on a single 
CPU these days, no matter how studly it is.


-Dan
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[Asterisk-Users] Configuring Meeting Room from Asterisk Manager API

[Somesh S Shanbhag]

Hi All,  I want to do a three-party conferencing using manager api.  But I found out from the asterisk-users list that I *MUST* use  the meeting room concept.  I wanted to know wheather meeting room can be configured dynamically?  on the fly? Otherwise, configuring meeting room statically is not scalable.  Thanks Regards, Somesh S. Shanbhag 
	
	
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RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion

[Wai Wu]

I don't think they are doing it with one Asterisk box. They did say "one rack 
of servers". Well, that might mean up to 50 computers if they are using blade 
servers.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Thursday, February 02, 2006 10:21 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system
rolloutquestion


[top-posting continued due to formatting sloth on my part]

So, then let me follow up with a few more comments:

1) I will make some assumptions from your note:

a) Asterisk is currently capable (unless something has "broken" 
recently) of handling 2500 SIP-SIP calls with no transcoding, 
including RTP sessions, if on an operating system and hardware that 
is appropriately configured.  This puts to rest some who have claimed 
that 5000 channels is "impossible" with Asterisk regardless of 
platform, at least according to Signate.

b) It is unclear if other channel drivers (IAX and Zaptel, 
specifically) have had any testing with significant numbers of 
channels.

c) It is unclear if anything other than pure RTP passthrough is 
viable in these configurations.  Maybe IVR causes collapse.  ?


2) Still no claims or comments on the specific testing methods, or on 
methodology.  I'm left still scratching my head as to if this is 
actually possible, since there is no specific claim that can be 
verified.  While I hope that your system can do those numbers (it 
would help me greatly in the future!) I can't say that I'm confident 
yet.  I'll follow up in private email for further discussion.


3) Nobody else has thus far taken the bait and made any comments 
about their systems. I appreciate Signate's comments; they seem to be 
the only ones to publicly claim large-scale throughput using Asterisk 
in a public forum.  Most other people who claim thousands or even 
high hundreds of connections do so offhand, without responding to 
second questions when I raise my figurative eyebrows.


4) There are still no notes on other problems with scale here.  I've 
had systems with several hundred simultaneous SIP connections, but 
"sip show channels" sure does start to take a while.  What _other_ 
problems crop up, but don't necessarily cause a "failure" condition?


5) I will agree that most SIP testing systems are currently too 
pricey.  I would love to find a well-connected network that rents out 
a few of the better-known SIP testing tools to beat on Asterisk 
installations in remote places for short periods of time.   But this 
has always been the case... test gear is a small market, and 
expensive.  Just look at the MSRP of new high-end HP Oscilloscopes if 
you want to get a picture of price-gouging.

JT



At 11:21 AM -0800 2/2/06, William Boehlke wrote:
>
>Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony
>Server 5000. The throughput has little to do with Asterisk and a lot to do
>with hardware design and operating system tuning. Our very minor code
>changes were returned to the project last year. 
>
>The benchmark we used to make that initial claim was flawed, however we have
>since replicated the throughput in a different way to save our marketing
>bacon.
>
>How we actually achieve the throughput is our intellectual property but we
>have a number of customers who are scaling towards and past that traffic
>level.  One of these days we hope to be able to justify the very large fee
>Hammer wants to extract from us to produce a third party verification.
>
>In production environments, of course, systems do more than switch calls. We
>think high volume system design using 32-bit systems of any kind is complex,
>and it's difficult to replicate the volumes without actual customer traffic
>- and by then it's too late. Where do you put voicemail? Where does the IVR
>reside?
>
>When someone needs to switch 5,000 calls with Class 5 services we would
>specify a rack of servers. The good news is that it is one rack, not three
>of them, but we need more than Asterisk alone, great though it is, to make
>everything work.
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
>Sent: Wednesday, February 01, 2006 9:33 PM
>To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
>Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout
>question
>
>
>>Signate sells a single server that can get you to the call volumes you
>need.
>>
>>Paul Mahler
>>[EMAIL PROTECTED]
>>www.signate.com
>>
>[snip]
>
>Past conversations on this topic have generated quite a bit of
>controversy within the Asterisk development community, both publicly
>here on the list forums as well as in quite a few more quiet
>discussions with people who often do not post but have extensive
>operational experiences with Asterisk (most of whom monitor the -dev
>list and whose replies will be suited to that audience.)
>
>The sub

Re: [Asterisk-Users] Zhone channel Banks

[Matt Florell]

I wrote a perl script to auto-config the zplex 10 as a B8ZS/ESF 24
port channelbank:

http://astguiclient.sourceforge.net/experimental_code/Zhone_zplex_24s_program.pl

Saved me many hours of frustrating monotonous configing/

MATT---


On 2/2/06, Jon Pounder <[EMAIL PROTECTED]> wrote:
>
> > I've got a Zhone 24 port FXS to configure.  The configuration is
> > beyond stupid.  The people that designed this unit should be chased
> > down and fired.
>
> its not exactly straightforward is it ?
>
> I don't have a file for that one, I have an 8/16, but one tip I can offer
> is when you get a file and try to upload it, use a super low speed setting
> in your terminal program (intercharacter delay), otherwise the zhone
> misses characters, silently errors and you pull your hair even more trying
> to figure out why if you enter the line manually it works but as part of a
> file upload it fails.
>
> ie : make it so slow you could almost type faster
>
>
> >
> > I'm going around in circles frigging with all the options.  Does
> > anyone have a config file for this unit that I can use as a starting
> > point?
> >
> > -bill
> > [EMAIL PROTECTED]
> >
> > ___
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> >
>
>
> Jon Pounder
>
>_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
> _/_/_/  _/  _/ _/_/_/  _/  _/_/
>_/_/  _/_/  _/ _/_/  _/_/  _/
> _/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/
>
>
> Inline Internet Systems Inc.
> Thorold, Ontario, Canada
>
> Tools to Power Your e-Business Solutions
> www.inline.net
> www.ihtml.com
> www.ihtmlmerchant.com
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RE: [Asterisk-Users] PRI Presentation Restricted bit honored?

[Nabeel Jafferali]

> Hi.  I'm wondering if it is possible to make asterisk honor the
> Presentation Restricted bit on incoming PRI calls.

I'm guessing you have to make your dialplan remove the CLI based on the
${CALLINGPRES} variable.

Nabeel

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RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[Nabeel Jafferali]

> How can I send the hook flash to the x100P card to switch to the call
> coming in from the PSTN?

http://www.voip-info.org/wiki-Asterisk+cmd+Flash

Scroll down to "Re: X100P + Call-Waiting how-to"

Enjoy.

Nabeel

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Re: [Asterisk-Users] Zhone channel Banks

[Jon Pounder]

> I've got a Zhone 24 port FXS to configure.  The configuration is
> beyond stupid.  The people that designed this unit should be chased
> down and fired.

its not exactly straightforward is it ?

I don't have a file for that one, I have an 8/16, but one tip I can offer
is when you get a file and try to upload it, use a super low speed setting
in your terminal program (intercharacter delay), otherwise the zhone
misses characters, silently errors and you pull your hair even more trying
to figure out why if you enter the line manually it works but as part of a
file upload it fails.

ie : make it so slow you could almost type faster


>
> I'm going around in circles frigging with all the options.  Does
> anyone have a config file for this unit that I can use as a starting
> point?
>
> -bill
> [EMAIL PROTECTED]
>
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Jon Pounder

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Re: [Asterisk-Users] username not stabled? * DO NOT USE USERNAME for locally attached phones!!!

[Ronald Wiplinger]

Olle E Johansson wrote:

Chris A. Icide wrote:

Ronald Wiplinger wrote:





601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf
621 and 626 are in Real-time sip_buddies

621 and 626 changes username back from name to number (name) in the
database, and never shows it in "sip show peer"

615 changed username "Ronald office" to 615, although no change in
sip.conf

Did anybody else experienced that?

*CLI> show version
Asterisk SVN-trunk-r8447M built by root @ vpbx on a x86_64 running
Linux on 2006-01-25 15:33:01 UTC




DO NOT USE USERNAME!

That feild is really something used together with defaultIP when we 
have *no* registration.


* When we have a registration, we use whatever the phone tells us in 
the Contact: header, which is a SIP requirement. That is why we change 
it to reflect the known contact, provided by the phone.


* We never change the device name of the phone, just the address we 
use to communicate with the phone.


* This name is not used for authentication by a phone, it is not the 
username you configure the phone with.


* In most cases, there is no need to use the "username=" option in 
sip.conf for phones.


I will soon change this setting to "defaultuser" to make it even more 
obvious that this is not anything you need to set. The phone name is 
whatever you have between the square brackets in sip.conf, both for 
users and peers.


I will check into the realtime code as well, to make sure we are 
changing the proper field in the database.




Olle,

thanks for explaining it.
Please think in my way for a moment and hopefully we get something usefully.

I have many remote users, and to make the life easy I use their existing 
e.164 phone number. That way nobody of my users need to think what 
number the other party has on our system or PSTN,   
As more users you get, as less you will remember their "name". Therefore 
I tried to use the field username for this help.
It is easy to remember 49 is the (only) guy in Germany and 8621 is our 
Shanghai person, but most of the people are in 8862 (Taipei)


Is there another solution ?


bye

Ronald Wiplinger
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Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection

[Jolly M. Recto]

hi,

I am just like yours a Satellite HUB operator that providing a voip. 
Before i am providing a ptp h323 with g723 codec boxes ranging 2 to 4 
port at 64kbps upstream and shared 2mbps downstream and now i come up to 
put asterisk using a g723 and g729 of digium but not work for me because 
when the remote end used data like email, browsing the voice suffer the 
quality. I look back on my last experiment and i see that SER and 
OPENSER is much better solutions to provide just a voice and voicemail 
to call out. The 64kbps with Qos in the remote config will help a lot 
better to provide a 99 % satisfaction to the customer.


//jollyr
Cosmin Prund wrote:

At my HQ I’m instaling a 128kbit leased line connection, with 
guaranteed bandwidth to the Internet; The telco promises less then 20 
ms to the internet (to ronix.ro), no jitter and no packet loss. So I’m 
hoping for 40 ms times to net and small jitter J This is my „hub”.


For my „satelite” instalations I’m planning on grabing a connection 
from a different provider (as this telco provider is expensive) but 
I’m also considering a 64kbit leased line from the same provider, just 
in case my VoIP doesn’t work with the cheeper providers. My remote 
instalations will never have more then one „conversation” load, and 
this conversation would be ZAP to IAX or SIP. That is, the distant 
instalation will need to forward all calls coming in on the zap chanel 
to my HQ Asterisk. That’s all it will ever do J. I’m not sure 
„trunking” woud provide anything in this case as there will never be 
more then one concurent conversation from the remote * to my HQ *. I’m 
expecting IAX to provide better performance over SIP but not by much.


Considering my remote * instalations will never have more then one 
concurent conversation with my HQ and considering I can get a really 
good 64kbit line I guess I’m OK. As for my HQ, I’m sure I’m OK because 
I’ll get a 128 kbit line and I’ll be able to afford an upgrade to 
256kbit. I can actually go all the way to 2048 kbit, but that would no 
longer be economically viable.


So I’ll see how it goes, and I hope I’ll have the time to put in a 
comment on the „low bandwidth” wiki on voip-info.org.


Thanks to everyone for your help.



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *tim panton

*Sent:* Thursday, February 02, 2006 11:05 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth 
connection


On 2 Feb 2006, at 08:09, Cosmin Prund wrote:



Brrghhh: Bandwidth calculation is really foggy for me:

Using the calculator I’m getting about 23 kbps for both incoming and 
outgoing. What does this mean: Is a 64kbit link used at 71% capacity 
((23+23):64) or is it used at only 35% (23:64)? Will this vary over 
time (i.e: does the codec generate more then average data at times? 
How about less then average?)


It depends on what sort of link you have. Most links are full duplex 
(leased lines etc) which would be 35%


but some radio based links are half duplex which would be 71%

So for a 64k link you will (just about) get 3 729 calls.

If all the calls between are between the same two servers, you can use 
IAX trunking, which would push


you up to 5 calls. (What that tells you is that for 729 and gsm, the 
headers are as big as the data).


You talk about satellite stations, if you are going for a hub and 
spoke, you should put the hub


on the highest bandwidth link.



Thanks.



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Lith

*Sent:* Wednesday, February 01, 2006 11:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth 
connection


What codec is that using. G.729 will give you 10 calls at best over 
256k unless you're trunking with IAX2? I don't know anyone using lpc10...


Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads...

Regards
Rob

On 2/1/06, *Garth van Sittert* <[EMAIL PROTECTED] 
> wrote:


Hi Cosmin

You should be able to get about 12 simultaneous calls on a 128k line and
about 28 on a 256k line according to asteriskguru's bandwidth calculator
http://www.asteriskguru.com/tools/bandwidth_calculator.php.

Kind Regards
Garth

BitCo Data Communications
http://www.bitco.co.za

Cosmin Prund wrote:

Hello everyone, this is my first post to the list, so hello again.

We're a small company in Romania and we're trying to set up a really 

small
version of "call center". That is, we want to get a few land-lines 

from our

telco in different countys and "bridge" all calls to our HQ, in order to
make it cheeper for our clients to call us.

Unfortunatelly there's no ISP in our area that can deliver a broadband
connection for anything less th

[Asterisk-Users] Zhone channel Banks

[William Lloyd]

I've got a Zhone 24 port FXS to configure.  The configuration is  
beyond stupid.  The people that designed this unit should be chased  
down and fired.


I'm going around in circles frigging with all the options.  Does  
anyone have a config file for this unit that I can use as a starting  
point?


-bill
[EMAIL PROTECTED]

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RE: [Asterisk-Users] RE: 5,000 concurrent calls system rollout question

[John Todd]

[top-posting continued due to formatting sloth on my part]

So, then let me follow up with a few more comments:

1) I will make some assumptions from your note:

   a) Asterisk is currently capable (unless something has "broken" 
recently) of handling 2500 SIP-SIP calls with no transcoding, 
including RTP sessions, if on an operating system and hardware that 
is appropriately configured.  This puts to rest some who have claimed 
that 5000 channels is "impossible" with Asterisk regardless of 
platform, at least according to Signate.


   b) It is unclear if other channel drivers (IAX and Zaptel, 
specifically) have had any testing with significant numbers of 
channels.


   c) It is unclear if anything other than pure RTP passthrough is 
viable in these configurations.  Maybe IVR causes collapse.  ?



2) Still no claims or comments on the specific testing methods, or on 
methodology.  I'm left still scratching my head as to if this is 
actually possible, since there is no specific claim that can be 
verified.  While I hope that your system can do those numbers (it 
would help me greatly in the future!) I can't say that I'm confident 
yet.  I'll follow up in private email for further discussion.



3) Nobody else has thus far taken the bait and made any comments 
about their systems. I appreciate Signate's comments; they seem to be 
the only ones to publicly claim large-scale throughput using Asterisk 
in a public forum.  Most other people who claim thousands or even 
high hundreds of connections do so offhand, without responding to 
second questions when I raise my figurative eyebrows.



4) There are still no notes on other problems with scale here.  I've 
had systems with several hundred simultaneous SIP connections, but 
"sip show channels" sure does start to take a while.  What _other_ 
problems crop up, but don't necessarily cause a "failure" condition?



5) I will agree that most SIP testing systems are currently too 
pricey.  I would love to find a well-connected network that rents out 
a few of the better-known SIP testing tools to beat on Asterisk 
installations in remote places for short periods of time.   But this 
has always been the case... test gear is a small market, and 
expensive.  Just look at the MSRP of new high-end HP Oscilloscopes if 
you want to get a picture of price-gouging.


JT



At 11:21 AM -0800 2/2/06, William Boehlke wrote:


Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony
Server 5000. The throughput has little to do with Asterisk and a lot to do
with hardware design and operating system tuning. Our very minor code
changes were returned to the project last year. 


The benchmark we used to make that initial claim was flawed, however we have
since replicated the throughput in a different way to save our marketing
bacon.

How we actually achieve the throughput is our intellectual property but we
have a number of customers who are scaling towards and past that traffic
level.  One of these days we hope to be able to justify the very large fee
Hammer wants to extract from us to produce a third party verification.

In production environments, of course, systems do more than switch calls. We
think high volume system design using 32-bit systems of any kind is complex,
and it's difficult to replicate the volumes without actual customer traffic
- and by then it's too late. Where do you put voicemail? Where does the IVR
reside?

When someone needs to switch 5,000 calls with Class 5 services we would
specify a rack of servers. The good news is that it is one rack, not three
of them, but we need more than Asterisk alone, great though it is, to make
everything work.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, February 01, 2006 9:33 PM
To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout
question



Signate sells a single server that can get you to the call volumes you

need.


Paul Mahler
[EMAIL PROTECTED]
www.signate.com


[snip]

Past conversations on this topic have generated quite a bit of
controversy within the Asterisk development community, both publicly
here on the list forums as well as in quite a few more quiet
discussions with people who often do not post but have extensive
operational experiences with Asterisk (most of whom monitor the -dev
list and whose replies will be suited to that audience.)

The subject of load on a single chassis is still the most contentious
issue to date.  The Signate numbers of >5000 calls per chassis with
RTP are impressive, and there are others who claim more vaguely of
1000, 2000, or more calls into a single P4 server (with or without
media.)  Others say that there are inherent limits in the Asterisk
code which prevent more than ~500 calls from being processed with RTP
at any one time.  Opterons, FreeBSD, custom Linux loads, Solaris, and
ot

RE: [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?

[Michael Crown]

We do.

http://www.thevoipconnection.com

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, February 02, 2006 7:32 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Any Digium Supplier/reseller 
> accepts Paypal ?
> 
> 
> 
> Looking for a Digium Supplier/Reseller that accepts Paypal.
> 
> Thanks,
> 
> Isamar
> 
> 
> 

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[Asterisk-Users] PRI Presentation Restricted bit honored?

[Jim Gottlieb]

Hi.  I'm wondering if it is possible to make asterisk honor the
Presentation Restricted bit on incoming PRI calls.

Ideally I'd still like to see the number in the CDR but we can't let
users hear restricted numbers in their voicemail messages, etc.

The docs only seem to talk about outgoing calls.

Thanks...
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RE: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time

[Brian J. Murrell]

On Thu, 2006-02-02 at 21:22 -0500, [EMAIL PROTECTED] wrote:
> No, it will dial like a pass-through simultaneously to sip/iax
> extensions.

Right.  As I thought.

>   If you were to dial out to an analog port though, that
> would be different.
> 
> So in essence, you can have all the phones ringing at the same time.

Right.  My original question was about making Asterisk wait a number or
rings (or amount of time) before picking up a Zap line.  If the
rings/time were not reached while the line is still ringing, do nothing.

This allows a handset *on the same POTS line* as Asterisk to pick up and
Asterisk does nothing.  But if nobody picks up the POTS line (that
asterisk is on too) then it picks up.

I essentially want Asterisk to be an answering machine on the line.

b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell


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RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[kevin ling]

Hi,

In AAH, you can setup the "Incoming Calls" to ring your extension. Or to
ring extensions in a ring group. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith
Sent: Friday, February 03, 2006 6:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

OK I have looked everywhere and I can't get a clear understanding on how to
do this. If I have an x100P card connected to my home phone line and I am
receiving calls on my Cisco 7940 IP phone with a SIP firmware loaded on it.
How can I send the hook flash to the x100P card to switch to the call coming
in from the PSTN? I am using [EMAIL PROTECTED] 2.4. I can hear the call waiting
tone coming over the line but the phone doesn't recognize it.



Thanks



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RE: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount of time

[gw]

No, it will dial like a pass-through simultaneously to sip/iax
extensions.  If you were to dial out to an analog port though, that
would be different.

So in essence, you can have all the phones ringing at the same time.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian J.
Murrell
Sent: Thursday, February 02, 2006 7:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: delaying "answer" for a number of
ringsor an amount of time

On Thu, 2006-02-02 at 15:24 -0700, Bromont Quebec wrote:
> You need to take that "Wait" and "Answer" out of there
> 
> [from-pots]
> exten => s,1,Dial(SIP/brian&SIP/joe,30) exten => s,2,Voicemail(u2001) 
> exten => s,3,Hangup exten => s,102,Voicemail(b2001) exten => 
> s,103,Hangup exten => h,1,Hangup exten => i,1,Hangup

How does doing only that prevent Asterisk from picking up the POTS line
for a period of time (like 3 or 4 rings... or 10 seconds or so to give a
handset on the same POTS line an opportunity to pick it up first --
think answering machine)?  As I understand it removing the Wait and
Answer would cause Asterisk to pick the POTS line up right away and dial
brian and joe's phones with it.

Am I missing something?

b.

--
My other computer is your Microsoft Windows server.

Brian J. Murrell
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RE: [Asterisk-Users] routing question: multipath routing for SIP

[gw]

Yes, and, you will probably need a different 
method.
 
Are these t1's to the same provider?  Have you 
considered bonding the channels?
 
Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Script 
HeadSent: Thursday, February 02, 2006 6:32 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] routing question: multipath routing for 
SIP
I have two T1s and I'd like to split my SIP traffic over the two. 
I am looking at this:http://lartc.org/howto/lartc.rpdb.multiple-links.htmlwhat 
bothers me about it is the note "Note that balancing will not be 
perfect, as it is route based, and routes are cached. This means that routes to 
often-used sites will always be over the same provider.". If all my traffic goes 
to the same IP, which is a remote SER proxy, will my second T1 be utilized at 
all? Does anyone have any experiece with this?
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Re: [Asterisk-Users] limit sip sessions

[Miguel]

[EMAIL PROTECTED] wrote:


Shouldn't all sip users have different usernames?
(or am I missing some vital detail here?)

PaulH

Yes Paul, Im in El Salvador and my users like to "share" their 
usernames/passwords and the original owner doesnt like to pay for calls 
he hasnt made.

---
Miguel
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Re: [Asterisk-Users] Re: Contents of Asterisk-Users digest...

[Anthony Rodgers]

Are you kidding?

On Feb 2, 2006, at 6:47 AM, Will Velez wrote:





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RE: [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?

[Garrett Smith]

VoIPSupply.com accepts paypal...

Please contact me off list for more details.

GS

Garrett Smith
<[EMAIL PROTECTED]>
716-250-3408 Direct
716-903-9495 Cell
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 02, 2006 7:32 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?



Looking for a Digium Supplier/Reseller that accepts Paypal.

Thanks,

Isamar

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Re: [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount of time

[Brian J. Murrell]

On Thu, 2006-02-02 at 15:24 -0700, Bromont Quebec wrote:
> You need to take that "Wait" and "Answer" out of there
> 
> [from-pots]
> exten => s,1,Dial(SIP/brian&SIP/joe,30)
> exten => s,2,Voicemail(u2001)
> exten => s,3,Hangup
> exten => s,102,Voicemail(b2001)
> exten => s,103,Hangup
> exten => h,1,Hangup
> exten => i,1,Hangup

How does doing only that prevent Asterisk from picking up the POTS line
for a period of time (like 3 or 4 rings... or 10 seconds or so to give a
handset on the same POTS line an opportunity to pick it up first --
think answering machine)?  As I understand it removing the Wait and
Answer would cause Asterisk to pick the POTS line up right away and dial
brian and joe's phones with it.

Am I missing something?

b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell


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[Asterisk-Users] Any Digium Supplier/reseller accepts Paypal ?

[isamar]

Looking for a Digium Supplier/Reseller that accepts Paypal.

Thanks,

Isamar

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Re: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question

[Rafael Marconi]

One single machine cant do 5000 IP -> PSTN translations.


u can use some asterisk as frontend to do codec translation, from   
any codec to g711,
and deliver all calls to Asterisk with the Sangoma board in G711,  
Sangoma has a T3/E3 board


this exaple will reduce the load on the servers with pstn card.

If u want asterisk only to route 5000 calls, without rtp media, u  
dont need any of this .



Rafael Marconi
[EMAIL PROTECTED]
[EMAIL PROTECTED]



Em 02/02/2006, às 18:00, Wai Wu escreveu:

Isn't it ridiculous that Hammer charges an arm and a leg for any  
work they do. For systems as large as that one, we just setup a  
seperate one, connect them back to back and run automated script to  
burn it in.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of William
Boehlke
Sent: Thursday, February 02, 2006 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system  
rollout

question


How we actually achieve the throughput is our intellectual property  
but we
have a number of customers who are scaling towards and past that  
traffic
level.  One of these days we hope to be able to justify the very  
large fee

Hammer wants to extract from us to produce a third party verification.

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Re: [Asterisk-Users] meetme and dtmf

[Imran Ahmed]

> >> Step 3 The Iax client heve to send some other DTMF to the IVR.
> >
> >
> > How is the IVR still involved if the call has been transferred into a
> > conference room?
> >
> The IVR records the conversation between the other partecipant to the
> conference and wait '#' to stop recording and a '1'  to save the file.

may or may not work, try at your own risk:

1) Use a sip soft phone and set the dtmf mode = inband.
2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
info. (this is done so that asterisk ignores the inband dtmf on the
sip channel).
3) Design your dialplan such that asterisk should not depend on dtmf
from the sip call.
ex:

exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room
exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room.
exten xxx, 3, meetme(conference room)

once the sip call is in the conference then the ivr will detect dtmf
from the audio data. Note that before the sip call is in a conference
dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this
is not tested and only a test can confirm if it works.

drawbacks: dtmf will not be available to ivr until your call is in
conference. asterisk will never see any dtmf (which should be okay in
this specific case).
dtmf tones are not squelched so the other user in the conference will
hear dtmf tones.

Imran
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[Asterisk-Users] Receiving faxes with spandsp - strange problem

[Bartosz Piec]

Hello,

I'm trying to receive faxes with asterisk. My configuration is like this:

PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk

When I try to send a fax from PSTN fax I got the standard fax signal, 
Asterisk starts rxfax application and then call ends and there is no tif 
anywhere. On the fax display there is still one message: Calling...


Part of my extensions.conf:

[incoming]
  exten => 2933975,1,Goto(fax,2201,1)

[fax]
  exten => 2201,1,Macro(faxreceive)

[macro-faxreceive]
  exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
  exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
  exten => s,3,rxfax(${FAXFILE})
  exten => s,4,Congestion
  exten => s,103,SetVar([EMAIL PROTECTED])
  exten => s,104,Goto(3)

2933975 is the number on which asterisk should listen for new faxes and 
it works (in a way that rxfax is being run).


Info from asterisk console:

-- Executing Goto("SIP/62.111.174.65-f67004d0", "fax|2201|1") in 
new stack

-- Goto (fax,2201,1)
-- Executing Macro("SIP/62.111.174.65-f67004d0", "faxreceive") in 
new stack
-- Executing SetVar("SIP/62.111.174.65-f67004d0", 
"FAXFILE=/var/spool/asterisk/fax/1138819786.1.tif") in new stack
-- Executing DBget("SIP/62.111.174.65-f67004d0", 
"EMAILADDR=extensionemail/2201") in new stack

-- DBget: varname=EMAILADDR, family=extensionemail, key=2201
-- DBget: set variable EMAILADDR to [EMAIL PROTECTED]
-- Executing RxFAX("SIP/62.111.174.65-f67004d0", 
"/var/spool/asterisk/fax/1138819786.1.tif") in new stack


And that's all. Fax disconnects and there is no 
/var/spool/asterisk/fax/1138819786.1.tif file. What can be wrong?


I saw that people are using 'fax' in extensions.conf (exten => 
fax,1,rxfax(${FAXFILE})). When this can be used? It isn't working for me...


Maybe it is an issue with Cisco router configuration? Normal calls (not 
faxes) from PSTN lines work great...


I have installed asterisk 1.2.4 and spandsp 0.0.2pre23.

--
Best regards,
Bartosz Piec
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[Asterisk-Users] routing question: multipath routing for SIP

[Script Head]

I have two T1s and I'd like to split my SIP traffic over the two. I am looking at this:http://lartc.org/howto/lartc.rpdb.multiple-links.html
what bothers me about it is the note "Note that balancing will not be perfect, as it is route based, and routes
	  are cached. This means that routes to often-used sites will always
	  be over the same provider.". If all my traffic goes to the same IP, which is a remote SER proxy, will my second T1 be utilized at all? Does anyone have any experiece with this?
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RE: [Asterisk-Users] Asterisk video conference

[Rusty Shackleford]

Title: Message




  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Shain 
  LeeSent: Thursday, February 02, 2006 2:14 AMTo: 
  AsteriskSubject: [Asterisk-Users] Asterisk video 
  conference
  Hi , 
   
  I just wanted to know , how would be asterisk work with video calls ? 
  
  What are the hardware do we have to buy ? 
  Who are the providers  of particular harwares ? 
   
  Can we use video calls / video conferenceing in the LAN perfectly ? How 
  it would be depends on the WAN ? 
   
Asterisk's support for video over SIP is very rudimentary. Only to 
video codecs H.261, H.263, and H.263+ are supported, and even then, not 
very well. There is no support for dynamic negotiation of frame rates, etc. 
Queries to the -dev list, as to progress on these features were recently met 
with silence. We will be looking to jump into the project to support our own 
initiatives in the area of video in a few weeks. 
 
Until things change, your best bet for connecting SIP video phones is 
SER. 
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Re: [Asterisk-Users] limit sip sessions

[pdhales]

Shouldn't all sip users have different usernames?
(or am I missing some vital detail here?)

PaulH

- Original Message - 
From: "Miguel" <[EMAIL PROTECTED]>
To: "Asterisk User List" 
Sent: Friday, February 03, 2006 3:21 AM
Subject: [Asterisk-Users] limit sip sessions


> hi, is there a way to limit the sip session per username?. i mean, if i 
> have a sip session with asterisk using xxx as username, nobody can 
> register with that username until my session is terminated.
> thanks
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[Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[Chuck Smith]

OK I have looked everywhere and I can't get a clear understanding on how to
do this. If I have an x100P card connected to my home phone line and I am
receiving calls on my Cisco 7940 IP phone with a SIP firmware loaded on it.
How can I send the hook flash to the x100P card to switch to the call coming
in from the PSTN? I am using [EMAIL PROTECTED] 2.4. I can hear the call waiting
tone coming over the line but the phone doesn't recognize it.



Thanks



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[Asterisk-Users] Re: T.38 patch instruactions

[Ben Dinnerville]

Nudge

Ben Dinnerville wrote:

Hi All,

I am looking at pacthing up Asterisk to test T.38 passthrough and see
that there is a bug with some pacth / diff / new files that need to be
applied and compiled with Asterisk.

Can anyone provide a quick how-to on applying this patch?

Do all the files that are applied to the bug
(http://bugs.digium.com/view.php?id=5090) need to be applied?
Do they need to be applied in any particular order?

Any other tips / pointers / instructions on how to apply it? I have
searched this forum and read thru most of the notes for the bug
itself, but it is not exactly clear on what needs to be done.

Cheers,

Ben
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[Asterisk-Users] Re: delaying "answer" for a number of rings or an amount of time

[Bromont Quebec]

You need to take that "Wait" and "Answer" out of there

[from-pots]
exten => s,1,Dial(SIP/brian&SIP/joe,30)
exten => s,2,Voicemail(u2001)
exten => s,3,Hangup
exten => s,102,Voicemail(b2001)
exten => s,103,Hangup
exten => h,1,Hangup
exten => i,1,Hangup

>I want Asterisk to delay answering the POTS line >via a Wildcard (a Zap
>channel) by some period of time, either a number of >rings or just a
>number of seconds.
>
>I have tried this:
>
>[from-pots]
>exten => s,1,Wait(30)
>exten => s,n,Answer
>...
>exten => s,n,Dial(SIP/brian&SIP/joe,10,H)
>exten => s,n,Voicemail(u2001)
>exten => s,n,Hangup
>exten => s,103,Voicemail(u2001)
>exten => s,104,Hangup
>exten => h,1,Hangup
>exten => i,1,Hangup

>but that doesn't work.  It seems that as soon as >the first or second
>ring is detected, Asterisk has decided it will >answer the line, it just
>waits 30 seconds to do it.
>
>The problem I have is that the POTS line that >Asterisk is on is shared
>by handsets, and if somebody picks up a handset >soon enough, I don't
>want Asterisk to pick up the line.  Yeah, Asterisk >as an answering
>machine... until I can get at least one FXS >interface, anyway.

>Any ideas?

>b.


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RE: [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch

[Michael Collins]

The intent is not to start a flame war.  I simply want your viewpoints.
Feel free to email me offline with your comments on these subjects.
This is the place where the most informed Asterisk users come and thus
the place I chose to ask these questions.  If there is a better forum (I
hear that IRC is a possibility) then please advise.

In the meantime PLEASE don't flame anyone on this list.  If you want to
flame me please do so at [EMAIL PROTECTED]  Kevin is right - a two
week flame war is pointless.

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, February 02, 2006 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch

Michael Collins wrote:

> My questions to the Asterisk user community: are you at all concerned
about
> the complaints made by the "forkers?"  Or are they the ones who are
all
> forked up? ;)  What about Freeswitch?  Do you see that as a threat to
> Asterisk or simply as yet another competing product?  (After all,
there are
> other open source PBX projects available.)

This mailing list is not for politics or speculation on the merits of 
other projects. It is for discussion of the _use_ of Asterisk.

We do not need another two-week thread flaming everyone for their 
opinions. Don't start one.
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[Asterisk-Users] Re: 5, 000 concurrent calls system rollout question

[Mike Hammett]

Why is using ulaw or alaw an unlikely scenario?  I wouldn't use anything but 
ulaw\alaw.  The Bells can compete on price and will if they have to.  Where 
they CAN'T compete is quality.  If there were something better than 711, I'd 
offer that.  Well, there is 722, but not many things support it.




Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: <[EMAIL PROTECTED]>

To: 
Sent: Thursday, February 02, 2006 2:52 PM
Subject: Asterisk-Users Digest, Vol 19, Issue 19



Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."


Today's Topics:

  1. 5,000 concurrent calls system rollout question (Vic)
  2. How to handle "provider UNREACHABLE" in the dialplan?
 (Ronald Wiplinger)
  3. delaying "answer" for a number of rings or an amount of time
 (Brian J. Murrell)
  4. RE: 5,000 concurrent calls system rollout question
 (Michael Loftis)
  5. Re: 5,000 concurrent calls system rollout question
 (Michael Loftis)
  6. RE: Directed Call Pickup (Alex Barnes)
  7. Re: ISDN Eicon Diva Server V-BRI (Bartosz Jozwiak)
  8. RE: RE: 5, 000 concurrent calls system rollout question
 (William Boehlke)
  9. Re: ISDN Eicon Diva Server V-BRI (Jens Vagelpohl)
 10. Re: RE: Rewind MusicOnHold? (Dan Journo)
 11. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler)
 12. Agents, queues and zombies (Steve Rawlings)
 13. Fw: Agents, queues and zombies (Steve Rawlings)
 14. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler)
 15. Re: OT O'Reilly Asterisk TFOT (James Ronald)
 16. Re: Blocked Callerid ([EMAIL PROTECTED])
 17. Re: fax possibilities ([EMAIL PROTECTED])
 18. Re: RE: Rewind MusicOnHold? (Dan Journo)
 19. RE: OT O'Reilly Asterisk TFOT (Michael Collins)
 20. Re: OT O'Reilly Asterisk TFOT (Mark Phillips)
 21. RE: Anyone know a good ITSP in Canada that supports*?
 (Technical Support)
 22. Slightly OT: OpenPBX.org and Freeswitch (Michael Collins)


--

Message: 4
Date: Thu, 02 Feb 2006 12:17:03 -0700
From: Michael Loftis <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout
question
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed



--On February 3, 2006 3:56:21 AM +0900 Vic <[EMAIL PROTECTED]> wrote:



Hi, Joash,

thank you for your email. I was very relieved to hear that someone was
already doing this.

Can you please tell me more about your test? Why did you test it in a
first place?

For me, we need to come up with a system that needs to:

1. Handle 5,000 inbound SIP calls

2. offer IVR capability

3. Billing


You'd probably have to do some of your own work on this.  * makes 'CDR'
records but...well...you have to be careful how you do your scripts if you
want legible/useable CDRs.  There are some apps out there though that will
process and do some sort of billing for CDRs not sure of what where.



I thought that Asterisk would be up to the task, but, I am not sure as
to:

1. How many servers should I consider? 4? 10? Obviously, we will be
talking about probably core Xeon servers if this is what we need.


I'd say atleast 10maybe more...depending wholly on codec/transcoding
and amount of IVR scripting.



2. How hard would it be to implement?


Well...since your not well versed with *, and you're having trouble
understanding the difference between a protocol and a codec, it might be
really difficult for you.  You might want to farm it out.  There are a LOT
of * consultancies out there now.  If you can get up to speed on asterisk
pretty quickly and the various protocols and codecs then it's not
impossible.  The kicker is all the management/maintenance UI's and such.
But you might be able to use something like Signates sigMAN (never used it
or their products).



3. How bad is g729 quality?

4. IVR : if the call is SIP, can we do prompts without transcoding?


You're confusing protocols with codec's here again.  SIP is not a codec.
That said if your SIP client is using GSM and there are GSM prompts
available then the asterisk playback functions will use the GSM encoded
prompts.

Earlier you'd mentioned using POTS lines coming in/out.  If you're
gatewaying 5k POTS lines you'll need a lot of machines.  Because you'll be
doing  a lot of transcoding POTS is ulaw or alaw (depending on where in 
the

world you are) and unless you use (uncompressed) ulaw or alaw on your SIP
clients (very unlikely scenario) you'll be transcoding to/from GSM. G.729,
or whatever you're using.



Any other suggestions t

Re: SV: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time

[Brian J. Murrell]

On Thu, 2006-02-02 at 22:08 +0100, [EMAIL PROTECTED] wrote:
> http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html

OK.  The hardware is a wildcard though.  How does that answer apply?
Isn't it asterisk itself that is picking that call up?  Can't it delay
the pick up?  Maybe I am just misunderstanding your reference.

b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell


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[Asterisk-Users] callerid issue

[Dov Bigio]

Hello,
 
In my sip.conf I have each IP phones defined as 
follows.
 
[ext13]type=friendsecret=123qualify=yes[EMAIL PROTECTED]language=ptcontext=geralfromuser=ext13username=ext13host=dynamicdisallow=allallow=g729allow=ulaw
But, when I call from ext13 to ext12, the caller id 
that appears on Phone12 is ext12, and not ext13, so when the users wants to dial 
to a missed call number, his phone simply calls to himself, and not the the 
right caller.
 
I  tried to change parameters callerid and 
fromuser, with no success.
Even tried in extensions.conf to use SetCallerId, 
but nothing helped.
 
Am I missing something, or is there something 
wrong here?
 
Thank you
Dov
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Re: [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch

[Jean-Michel Hiver]

Michael Collins a écrit :

This is slightly OT in that it isn’t specifically *-related, but I was 
wondering what the members of the * user community felt about these 
two subjects. I’ve been perusing the OpenPBX.org mail list and the 
current hot topic is the fact that their project has come to a 
grinding halt. They are concerned that they don’t have enough people 
working on their project. They feel that * has improved since the fork 
but they still have the same complaints: the Asterisk core is a “pig,” 
the core is huge and messy with ugly code, and finally that Asterisk 
development is throttled by the “Digistapo.”


Please take your big hairy troll back home. Keep this for the IRC or 
your local LUG beer ranting.


--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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[Asterisk-Users] RE: Rewind MusicOnHold?

[Brent Torrenga]

Create a class of hold music for each channel? Then do a
Dial(SIP/blah,,m(Channel1))? Or something like that...

Also, I do not know if this behavior seems to vary from system to system,
but on mine it seems like each channel does get it's own music, that it is
not broadcast across al channels... Ymmv

--Brent

>In theory that is fine, however, when a call leaves the MoH, the MoH
doesn't
>stop playing. So when a second call is received, the MoH is still playing
>and therefore they dont hear the message for a few minutes.
>
>Dan
>
>
>On 02/02/06, Brent Torrenga <[EMAIL PROTECTED]> wrote:
>>
>> The native MOH type will, unless set to random=yes, play the music files
>> in
>> the same order as they appear with an ls of the directory. (someone,
>> anyone,
>> back me up here?)
>>
>> I would place the greeting in the same MOH class as your actual music,
and
>> name the file of the greeting something "less" than the filename of the
>> music file. Additionally, to avoid repeating the greeting, should the
>> music
>> file play all the way through before an answer, you may want to make
>> additional copies of the music file, named something "greater" than the
>> greeting file.
>>
>> Just a thought, never tried to do it myself.

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RE: [Asterisk-Users] Asterisk on laptop connected to POTS line

[asterisk]

On Thu, 2 Feb 2006, Jonathan k. Creasy wrote:

The Grandstream ATA (480 I think...) does this and usually costs less
than the Sipura. It has 1 FXS and 1 FXO.


and both the sipura and grandstream have very poor echo cancellation.
not exactly the thing you want to be showing off to prospective customers.

-Dan
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Re: [Asterisk-Users] RE: Rewind MusicOnHold?

[Dan Journo]

Ok, i feel like im getting somewhere but i need a little help.
 
Asterisk displays this when its loading:-
[res_musiconhold.so] => (Music On Hold Resource)  == Registered application 'MusicOnHold'  == Registered application 'WaitMusicOnHold'  == Registered application 'SetMusicOnHold'  == Registered application 'StartMusicOnHold'
  == Registered application 'StopMusicOnHold'  == Parsing '/etc/asterisk/musiconhold.conf': Found 
But what does the StopMusicOnHold application do? and how can I use it?
 
Thanks
Dan 
On 02/02/06, Dan Journo <[EMAIL PROTECTED]> wrote:

Is there a command to playback() a file while continuing to process the dialplan?
 
Or, is there a way to start the MoH playback while continuing to process the dialplan? 
Should I be looking at the queues?
 
Dan 

On 02/02/06, Dan Journo <[EMAIL PROTECTED]
> wrote: 

In theory that is fine, however, when a call leaves the MoH, the MoH doesn't stop playing. So when a second call is received, the MoH is still playing and therefore they dont hear the message for a few minutes.

 
Dan 

On 02/02/06, Brent Torrenga <[EMAIL PROTECTED] 
> wrote: 
The native MOH type will, unless set to random=yes, play the music files inthe same order as they appear with an ls of the directory. (someone, anyone, 
back me up here?)I would place the greeting in the same MOH class as your actual music, andname the file of the greeting something "less" than the filename of themusic file. Additionally, to avoid repeating the greeting, should the music 
file play all the way through before an answer, you may want to makeadditional copies of the music file, named something "greater" than thegreeting file.Just a thought, never tried to do it myself. 
Sincerely,Brent A. Torrenga[EMAIL PROTECTED]Torrenga Engineering, Inc. 
907 Ridge RoadMunster, Indiana 46321-1771219.836.8918x325 Voice219.836.1138 Facsimilewww.torrenga.com 
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SV: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time

[jan.sarin]

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html


-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] genom Brian J. Murrell
Skickat: to 2006-02-02 20:14
Till: asterisk-users@lists.digium.com
Ämne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of 
time
 
I want Asterisk to delay answering the POTS line via a Wildcard (a Zap
channel) by some period of time, either a number of rings or just a
number of seconds.

I have tried this:

[from-pots]
exten => s,1,Wait(30)
exten => s,n,Answer
...
exten => s,n,Dial(SIP/brian&SIP/joe,10,H)
exten => s,n,Voicemail(u2001)
exten => s,n,Hangup
exten => s,103,Voicemail(u2001)
exten => s,104,Hangup
exten => h,1,Hangup
exten => i,1,Hangup

but that doesn't work.  It seems that as soon as the first or second
ring is detected, Asterisk has decided it will answer the line, it just
waits 30 seconds to do it.

The problem I have is that the POTS line that Asterisk is on is shared
by handsets, and if somebody picks up a handset soon enough, I don't
want Asterisk to pick up the line.  Yeah, Asterisk as an answering
machine... until I can get at least one FXS interface, anyway.

Any ideas?

b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell

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Re: [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch

[Kevin P. Fleming]

Michael Collins wrote:


My questions to the Asterisk user community: are you at all concerned about
the complaints made by the "forkers?"  Or are they the ones who are all
forked up? ;)  What about Freeswitch?  Do you see that as a threat to
Asterisk or simply as yet another competing product?  (After all, there are
other open source PBX projects available.)


This mailing list is not for politics or speculation on the merits of 
other projects. It is for discussion of the _use_ of Asterisk.


We do not need another two-week thread flaming everyone for their 
opinions. Don't start one.

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[Asterisk-Users] stream file 16k sample and 16 bit data

[Jerry Geis]

All,

I tried to playback a wave file what was 16k sample and 16 bit data it 
does not play...


using sox...
I resampled to 8k and 16 bit data it plays but there is noticible loss 
in loudness.


I resampled again to get 8k 8it gsm and it plays but there is additional 
background hiss introduced.


Is there some way take the 16k 16 bit wave file directly so no 
degredation occurs?


Jerry
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RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question

[Wai Wu]

Isn't it ridiculous that Hammer charges an arm and a leg for any work they do. 
For systems as large as that one, we just setup a seperate one, connect them 
back to back and run automated script to burn it in.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of William
Boehlke
Sent: Thursday, February 02, 2006 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rollout
question


How we actually achieve the throughput is our intellectual property but we
have a number of customers who are scaling towards and past that traffic
level.  One of these days we hope to be able to justify the very large fee
Hammer wants to extract from us to produce a third party verification.

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[Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch

[Michael Collins]

This is slightly OT in that it isn’t specifically
*-related, but I was wondering what the members of the * user community felt
about these two subjects.  I’ve been perusing the OpenPBX.org mail
list and the current hot topic is the fact that their project has come to a
grinding halt.  They are concerned that they don’t have enough
people working on their project.  They feel that * has improved since the
fork but they still have the same complaints: the Asterisk core is a “pig,”
the core is huge and messy with ugly code, and finally that Asterisk
development is throttled by the “Digistapo.”

 

Some there have suggested that Freeswitch is going to be the
wave of the future.  They hyped it up pretty well.  However, after
visiting the Freeswitch site and signing up to get the source and docs, I’m
not all that impressed.  It’s a great idea but it is months away
from being a truly useful product.

 

My questions to the Asterisk user community: are you at all
concerned about the complaints made by the “forkers?”  Or are
they the ones who are all forked up? ;)  What about Freeswitch?  Do
you see that as a threat to Asterisk or simply as yet another competing
product?  (After all, there are other open source PBX projects available.)

 

I would really like to hear your comments, concerns and
viewpoints.

 

-MC






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RE: [Asterisk-Users] Anyone know a good ITSP in Canada that supports*?

[Technical Support]

We have tried 3 different ITSP's in Canada - 2 of which had terrible
performance (choppy voice, etc).  We've been with unlimitel.ca for a while
now and had good results. They have Toronto and Ottawa DIDs.  They also have
good tech support (small company means everyone knows what's going on, and
they bend over backwards to solve your problems).

As for USA, you will probably need another TISP.

MD

(Not affiliated with unlimitel either)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: Thursday, February 02, 2006 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone know a good ITSP in Canada that
supports*?

iBell just announced termination only to CA for I believe $0.0039 a minute.

--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:

> On Thursday 02 February 2006 07:39, hugolivude
> wrote:
> > I'm looking for a new Internet Telephony Service
> Provider for my company in
> > Canada to terminate calls from my Asterisk PBX. 
> Ideally I'd like DiDs in
> > Otawa, Toronto, NY & San Jose.  Anyone out ther
> who can help me with a
> > recommendation?
> 
> Unlimitel.ca.  CAD$0.011/min for origination and on-net termination.
> Excellent, and I mean *excellent* customer service.
> 
> Not affiliated, but a very happy customer.
> 
> -A.
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Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT

[Mark Phillips]

It does indeed.

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


James Ronald wrote:

Does the printed version have an index?
-- JR

Whilst it can be downloaded I find that a paper copy is easier to 
read. I bought it for that reason alone. I also find it's a usefull 
addition to my tool box. I can't always access the net whilst on site. 
If I get stuck doing something I can look it up in the book.


Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Dave Cotton wrote:


I went to the Linux Solutions exhibition in Paris yesterday, visited the
well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6
hours later there was only one left. It must say something, also it was
the English version. 


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RE: [Asterisk-Users] OT O'Reilly Asterisk TFOT

[Michael Collins]

It certainly does!
-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Ronald
Sent: Thursday, February 02, 2006 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT

Does the printed version have an index?
-- JR 

> Whilst it can be downloaded I find that a paper copy is easier to
read. 
> I bought it for that reason alone. I also find it's a usefull addition

> to my tool box. I can't always access the net whilst on site. If I get

> stuck doing something I can look it up in the book.
> 
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
> 
> 
> Dave Cotton wrote:
>> I went to the Linux Solutions exhibition in Paris yesterday, visited
the
>> well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6
>> hours later there was only one left. It must say something, also it
was
>> the English version. 
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Re: [Asterisk-Users] RE: Rewind MusicOnHold?

[Dan Journo]

Is there a command to playback() a file while continuing to process the dialplan?
 
Or, is there a way to start the MoH playback while continuing to process the dialplan? 
Should I be looking at the queues?
 
Dan 
On 02/02/06, Dan Journo <[EMAIL PROTECTED]> wrote:

In theory that is fine, however, when a call leaves the MoH, the MoH doesn't stop playing. So when a second call is received, the MoH is still playing and therefore they dont hear the message for a few minutes.

 
Dan 

On 02/02/06, Brent Torrenga <[EMAIL PROTECTED]
> wrote: 
The native MOH type will, unless set to random=yes, play the music files inthe same order as they appear with an ls of the directory. (someone, anyone, 
back me up here?)I would place the greeting in the same MOH class as your actual music, andname the file of the greeting something "less" than the filename of themusic file. Additionally, to avoid repeating the greeting, should the music 
file play all the way through before an answer, you may want to makeadditional copies of the music file, named something "greater" than thegreeting file.Just a thought, never tried to do it myself. 
Sincerely,Brent A. Torrenga[EMAIL PROTECTED]Torrenga Engineering, Inc.
907 Ridge RoadMunster, Indiana 46321-1771219.836.8918x325 Voice219.836.1138 Facsimilewww.torrenga.com
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Re: [Asterisk-Users] fax possibilities

[pdhales]

h - using hylafax or asterisk?

PaulH

- Original Message - 
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, February 02, 2006 10:13 AM
Subject: [Asterisk-Users] fax possibilities


> I am trying to set up a linux based faxing solution for a client, and
> have found that the modem they have (ancient dataplex external unit)
> just isn't up to the job. It talks to some remote fax machines but not
> others.
>
> A new external modem ranges from AUD$75 to AUD$400, which got me
> thinking of other possibilities...
>
> #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+)
> #2 Sipura SPA3000
> #3 Grandstream ATA488
>
> I assume there will be no problem getting #1 working as a fax modem, but
> what about #2 and #3? Has anyone done this before? Some brief googling
> shows that it is possible, but not that it has been done...
>
> James
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Re: [Asterisk-Users] Blocked Callerid

[pdhales]

Do you mean 1-800 number? 
 
I don't really know the answer - I will have to ask 
next time I visit.
 
PaulH
 

  - Original Message - 
  From: 
  Joe Pukepail 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, February 02, 2006 7:47 
  AM
  Subject: Re: [Asterisk-Users] Blocked 
  Callerid
  Do they have an 800 number?  If so perhaps their 800 
  number provider is doing it via DTMF.  Search around on the internet, I 
  believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or 
  perhaps reversed). 
  On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: 
  
  
I have been discussing an asterisk solution 
with a company that has a custom written dialogic based 
solution.
 
The issue is that their dialogic solution can 
read callerid from incoming calls, even if the callerid is 
blocked.
I have read before that Asterisk can do this, 
and they want me to make sure that their new system will be able to do 
this.
 
A quick poke around inside the zaptel source 
code was unproductive...
 
Any ideas?
 
PaulH
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Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT

[James Ronald]

Does the printed version have an index?
-- JR 

Whilst it can be downloaded I find that a paper copy is easier to read. 
I bought it for that reason alone. I also find it's a usefull addition 
to my tool box. I can't always access the net whilst on site. If I get 
stuck doing something I can look it up in the book.


Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Dave Cotton wrote:

I went to the Linux Solutions exhibition in Paris yesterday, visited the
well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6
hours later there was only one left. It must say something, also it was
the English version. 

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Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Armin Schindler]

On Thu, 2 Feb 2006, Jens Vagelpohl wrote:
> > I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and
> > run with chan_capi.
> > Is anybody using that card ? Would appreciate any feedback.
> 
> Card works great with chan_capi-cm from sourceforge. Don't overlook the small
> print: the V series cards don't do any fax. The standard Diva Server BRI cards
> do (I assume).

Yes, the non-V cards include Fax/Modem and Voice (and of course other 
protocols like X.75, HDLC, V.110, ...) as well as Line-Interconnect
(channel-to-channel, card-to-card bridging).
 
> The V cards are branded as optimized for voice, but at this point I'm not sure
> what the advantage really is in a Asterisk system.

For example the echo canceller. But as mentioned I will also implement the 
RTP (Eicon DIVA Server cards can work directly with RTP packets and do
the codec/anti-jitter stuff itself).
So when finished, the SIP phones will send their RTP packets to the DIVA 
card instead of Asterisk.

Armin
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[Asterisk-Users] Fw: Agents, queues and zombies

[Steve Rawlings]

Sorry,

Forgot to mention, I'm running Asterisk 1.2.3, zaptel and libpri 1.2.2, 
addons 1.2.1.


Steve

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[Asterisk-Users] Agents, queues and zombies

[Steve Rawlings]

Hi all,

Have been experimenting with agents and queues instead of placing calls 
direct to a user's phone extension, but I've run into problems with calls to 
both the agent and the extension which creates a zombie and double records 
calls abandoned etc.  We're using a unique queue for each agent (only a 
handful of users) to try and get some agent/queue information to see what 
the agents have been doing, but for some reason each call rings both the 
agent and the extension, the extension drops once answered but I get two cdr 
records, if the call is abandoned I get two calls added to the total when I 
check with 'show queues'.  Any ideas how I can avoid this, I've searched the 
wiki with little joy, have I misssed something simple?


Steve




I've created the agents,
agents.conf
..
agent => 366,,Richard
agent => 300,,Jamie

created the queues
queues.conf
..
[Q204]
member => Agent/366

[Q205]
member => Agent/300

Logged them in with
exten => 555,1,AgentCallbackLogin

This requests an agent ID, no password and the extension to ring for the 
calls and lets me login anyone to any phone.  For the test I logged in agent 
300 and used extension 205.  The testing dialplan for incoming calls over 
our PRI goes something like -


extensions.conf
..
exten => 205,1,Answer
exten => 205,2,Ringback
exten => 205,3,Wait(2)
exten => 205,4,Queue(Q205)

The call gets answered and the log shows it's
VERBOSE[10393] logger.c: -- Executing Queue("Zap/32-1", "Q205") in new stack
Feb 2 13:47:20 DEBUG[10393] channel.c: Prodding channel 'Zap/32-1'

Feb 2 13:47:20 VERBOSE[10393] logger.c: -- Started music on hold, class 
'default', on channel 'Zap/32-1'


Feb 2 13:47:20 DEBUG[10393] channel.c: Scheduling timer at 160 sample 
intervals


Feb 2 13:47:20 VERBOSE[10393] logger.c: -- outgoing agentcall, to agent 
'300', on 'Local/[EMAIL PROTECTED],1'


Feb 2 13:47:20 VERBOSE[10393] logger.c: -- Called Agent/300

Feb 2 13:47:20 VERBOSE[10395] logger.c: -- Executing 
Dial("Local/[EMAIL PROTECTED],2", "SIP/205|30|tT") in new stack


Feb 2 13:47:20 DEBUG[10395] chan_sip.c: Outgoing Call for 205

Feb 2 13:47:20 VERBOSE[10395] logger.c: -- Called 205

Feb 2 13:47:20 DEBUG[1209] chan_sip.c: (Provisional) Stopping retransmission 
(but retaining packet) on '[EMAIL PROTECTED]' 
Request 102: Found


Feb 2 13:47:20 DEBUG[1209] chan_sip.c: (Provisional) Stopping retransmission 
(but retaining packet) on '[EMAIL PROTECTED]' 
Request 102: Found


Feb 2 13:47:20 VERBOSE[10395] logger.c: -- SIP/205-e065 is ringing

Feb 2 13:47:20 VERBOSE[10393] logger.c: -- Agent/300 is ringing

Feb 2 13:47:20 DEBUG[10393] channel.c: Generator got voice, switching to 
phase locked mode


Feb 2 13:47:22 DEBUG[1209] chan_sip.c: build_route: Contact hop: 



Feb 2 13:47:22 VERBOSE[10395] logger.c: -- SIP/205-e065 answered 
Local/[EMAIL PROTECTED],2


Feb 2 13:47:22 DEBUG[10393] app_queue.c: Dunno what to do with control 
type -1


Feb 2 13:47:22 VERBOSE[10393] logger.c: -- Agent/300 answered Zap/32-1

Feb 2 13:47:22 DEBUG[10393] chan_zap.c: Set option TONE VERIFY, mode: 
MUTECONF(1) on Zap/32-1


Feb 2 13:47:22 VERBOSE[10393] logger.c: -- Stopped music on hold on Zap/32-1

Feb 2 13:47:22 DEBUG[10393] channel.c: Scheduling timer at 0 sample 
intervals


Feb 2 13:47:22 DEBUG[10395] channel.c: Planning to masquerade channel 
SIP/205-e065 into the structure of Local/[EMAIL PROTECTED],1


Feb 2 13:47:22 DEBUG[10395] channel.c: Done planning to masquerade channel 
SIP/205-e065 into the structure of Local/[EMAIL PROTECTED],1


Feb 2 13:47:22 DEBUG[10393] channel.c: Got clone lock for masquerade on 
'SIP/205-e065' at 0x8801ee4


Feb 2 13:47:22 DEBUG[10395] chan_local.c: Not posting to queue since already 
masked on 'Local/[EMAIL PROTECTED],2'


Feb 2 13:47:22 DEBUG[10393] channel.c: Putting channel SIP/205-e065 in 64/64 
formats


Feb 2 13:47:22 DEBUG[10393] channel.c: Released clone lock on 
'Local/[EMAIL PROTECTED],1'


Feb 2 13:47:22 DEBUG[10393] channel.c: Done Masquerading SIP/205-e065 (6)

Feb 2 13:47:22 DEBUG[10393] chan_agent.c: Bridge on 'SIP/205-e065' being set 
to 'Agent/300' (3)


Feb 2 13:47:22 DEBUG[10393] chan_agent.c: Native formats changing from 64 to 
8


Feb 2 13:47:22 DEBUG[10393] chan_agent.c: Resetting read to 64 and write to 
64


Feb 2 13:47:22 DEBUG[10395] channel.c: Bridge stops because we're zombie or 
need a soft hangup: c0=Local/[EMAIL PROTECTED],2, 
c1=Local/[EMAIL PROTECTED],1, flags: No,No,Yes,Yes


Feb 2 13:47:22 DEBUG[10395] channel.c: Bridge stops bridging channels 
Local/[EMAIL PROTECTED],2 and Local/[EMAIL PROTECTED],1


Feb 2 13:47:22 DEBUG[10395] app_dial.c: Exiting with DIALSTATUS=ANSWER.

Feb 2 13:47:22 VERBOSE[10395] logger.c: == Spawn extension 
(macro-stdextn-sip, xindex, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' 
in macro 'stdextn-sip'


Feb 2 13:47:22 VERBOSE[10395] logger.c: == Spawn extension (default, 205, 1) 
exited non-zero on 'Local/[EMAIL PROTECTED],2'


Feb 2 13:47:22 DEBUG[1140] channel.c: Avoiding initial deadlock 

Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Armin Schindler]

On Thu, 2 Feb 2006, Bartosz Jozwiak wrote:
> > On Thu, 2 Feb 2006, Bartosz Jozwiak wrote:
> > > Dear all,
> > > 
> > > I'm planning to buying Eicon Diva Server V-BRI for my asterisk server
> > > and run
> > > with chan_capi.
> > > Is anybody using that card ? Would appreciate any feedback.
> > 
> > I have the non-V version of that card multiple times in use with
> > perfect results.
> > Do you need specific information?
> > 
> 
> Do you have it working with chan_capi ?

Sure, chan_capi-cm. Soon I will introduce CAPI-RTP to chan_capi
for DIVA Server Cards. (I have it already running for tests with OpenPBX).
 
> So I guess the V version is going to work too.

Yes, of course.

> I see that V version is even cheaper, wonder why.

The V cards are (V)oice only. Other features like Fax are not available.
The Eicon website http://www.eicon.com should provide this information
(I don't have all this in mind).

Armin
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Re: [Asterisk-Users] RE: Rewind MusicOnHold?

[Dan Journo]

In theory that is fine, however, when a call leaves the MoH, the MoH doesn't stop playing. So when a second call is received, the MoH is still playing and therefore they dont hear the message for a few minutes.
 
Dan 
On 02/02/06, Brent Torrenga <[EMAIL PROTECTED]> wrote:
The native MOH type will, unless set to random=yes, play the music files inthe same order as they appear with an ls of the directory. (someone, anyone,
back me up here?)I would place the greeting in the same MOH class as your actual music, andname the file of the greeting something "less" than the filename of themusic file. Additionally, to avoid repeating the greeting, should the music
file play all the way through before an answer, you may want to makeadditional copies of the music file, named something "greater" than thegreeting file.Just a thought, never tried to do it myself.
Sincerely,Brent A. Torrenga[EMAIL PROTECTED]Torrenga Engineering, Inc.907 Ridge RoadMunster, Indiana 46321-1771219.836.8918x325
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Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Jens Vagelpohl]

On 2 Feb 2006, at 18:25, Bartosz Jozwiak wrote:


Dear all,

I'm planning to buying Eicon Diva Server V-BRI for my asterisk  
server and run with chan_capi.

Is anybody using that card ? Would appreciate any feedback.


Card works great with chan_capi-cm from sourceforge. Don't overlook  
the small print: the V series cards don't do any fax. The standard  
Diva Server BRI cards do (I assume).


The V cards are branded as optimized for voice, but at this point I'm  
not sure what the advantage really is in a Asterisk system. Had I  
known about the Fax situation I would have gone for the standard  
card, which I believe is a little more expensive.


jens

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RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout question

[William Boehlke]

Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony
Server 5000. The throughput has little to do with Asterisk and a lot to do
with hardware design and operating system tuning. Our very minor code
changes were returned to the project last year.  

The benchmark we used to make that initial claim was flawed, however we have
since replicated the throughput in a different way to save our marketing
bacon. 

How we actually achieve the throughput is our intellectual property but we
have a number of customers who are scaling towards and past that traffic
level.  One of these days we hope to be able to justify the very large fee
Hammer wants to extract from us to produce a third party verification.

In production environments, of course, systems do more than switch calls. We
think high volume system design using 32-bit systems of any kind is complex,
and it's difficult to replicate the volumes without actual customer traffic
- and by then it's too late. Where do you put voicemail? Where does the IVR
reside? 

When someone needs to switch 5,000 calls with Class 5 services we would
specify a rack of servers. The good news is that it is one rack, not three
of them, but we need more than Asterisk alone, great though it is, to make
everything work. 



  





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, February 01, 2006 9:33 PM
To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout
question


>Signate sells a single server that can get you to the call volumes you
need.
>
>Paul Mahler
>[EMAIL PROTECTED]
>www.signate.com
>
[snip]

Past conversations on this topic have generated quite a bit of 
controversy within the Asterisk development community, both publicly 
here on the list forums as well as in quite a few more quiet 
discussions with people who often do not post but have extensive 
operational experiences with Asterisk (most of whom monitor the -dev 
list and whose replies will be suited to that audience.)

The subject of load on a single chassis is still the most contentious 
issue to date.  The Signate numbers of >5000 calls per chassis with 
RTP are impressive, and there are others who claim more vaguely of 
1000, 2000, or more calls into a single P4 server (with or without 
media.)  Others say that there are inherent limits in the Asterisk 
code which prevent more than ~500 calls from being processed with RTP 
at any one time.  Opterons, FreeBSD, custom Linux loads, Solaris, and 
other operating systems or hardware have been offered as the magic 
bullets to increase call volumes.  Who knows? (1)  I will say that 
extraordinary claims demand extraordinary evidence, which has been 
pretty thin.  I believe that most large call processing facilities 
still run on distributed systems of some type, as was described in 
the primary thread of this discussion on -users. (2)

I know that there are some projects towards testing Asterisk more 
rigorously to determine these numbers.  However, I would suggest that 
the community at large could benefit from a more open examination of 
high-end system claims immediately than these (better) long-term 
tests which are progressing slowly (if at all.)  Let's just look at 
the "maximum" numbers.  Running a big system? Selling a big system? 
Tell us about it, in detail.  What are the limits that have been hit? 
Be specific.  I keep seeing hand-waving, but no programmers have come 
forward to say "It won't work because of the way X is implemented in 
the file blah.c or libFOO."

To make a bad analogy:  I don't want to see the street rods; I just 
want to see the top-fuel, rocket-powered dragsters on the line.  Any 
takers?  It sounds like Signate has a contender, but quite a few 
people have said that it's impossible without serious modifications 
to the code.  Others have claimed (publicly or privately) that they 
can match those numbers on different hardware.

Here are the criteria:
   - Any O/S
   - An unmodified version of Asterisk from SVN (or CVS)
   OR patches must be available for inspection, as per the GPL
   OR you must be a Digium license-holder (patches can be secret)
   - All calls are IAX2 or SIP (both in and out)
   - No transcoding of any type is required
   - All calls are G.711, 20ms OR 30ms packet size

Documentation:
   - All O/S documentation, kernel tricks, modules, hacks, patches, or 
configuration elements should be documented, but proprietary 
information need not be divulged if that is deemed "secret"
   - Testing method must be reasonably documented
   - Dialplans must be included
   - SIP.conf files must be included
   - All hardware must be fully described (part numbers required)

TEST #1:
All media must be handled by the server.  This is for both legs of 
the call.  The "canreinvite=no" for SIP and "notransfer=yes" in IAX2 
must be set for al

Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Bartosz Jozwiak]

On Thu, 2 Feb 2006, Bartosz Jozwiak wrote:

Dear all,

I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and 
run

with chan_capi.
Is anybody using that card ? Would appreciate any feedback.


I have the non-V version of that card multiple times in use with
perfect results.
Do you need specific information?



Do you have it working with chan_capi ?

So I guess the V version is going to work too.
I see that V version is even cheaper, wonder why.

Bartosz 


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RE: [Asterisk-Users] Directed Call Pickup

[Alex Barnes]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Garth van Sittert
> Sent: 02 February 2006 16:47
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Directed Call Pickup
> 
> Hi All
> 
> I am having problems with Directed Call Pickup in Asterisk 1.2.1
> 
> If extension 100 is ringing, a user at another extension is supposed
to
> be able to dial *8100 and pickup the call to 100.  It isn't working
for
> me and I cannot figure out why.
> 
> I have in features.conf:
> 
> pickupexten = *8
> 


I am running 1.2.1 and works for me.


exten => _86.,1,Macro(directedPickup) ;  Direct Pickup

[macro-directedPickup]
exten => s,1,Pickup(${MACRO_EXTEN:2});


Remember that the *8 in your features.conf has nothing to do with direct
pickup.  So in your case try replacing _86. with _*8. but I don't know
if that will cause problems.

HTH

Alex

---
Alex Barnes
Engineering Support
Ubiquity Software
---


Information contained in this e-mail and any attachments are intended for the 
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Software Corporation.  All unauthorized use, disclosure or distribution is 
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

[Michael Loftis]

--On February 3, 2006 4:07:05 AM +0900 Vic <[EMAIL PROTECTED]> wrote:



Hi,

several of your mentioned signant as a viable option.

Has anyone ever used them? Are there any reviews for their products?

Did they just put together a lot of Asterisks into a large scale PC? (I
am still struggling with the concept)


Well I've nebver used it but any single box solution is going to have to 
have custom hardware and some custom code in asterisk or asterisk channel 
module to run it.  A PC can't do echo cancellation on 5k channels.  Can't 
do codec on 5k channels.  It might be able to do (light/simple/short) IVR 
on 5k channels though.




Thanks,

Vic




--
"Genius might be described as a supreme capacity for getting its possessors
into trouble of all kinds."
-- Samuel Butler
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RE: [Asterisk-Users] 5,000 concurrent calls system rollout question

[Michael Loftis]

--On February 3, 2006 3:56:21 AM +0900 Vic <[EMAIL PROTECTED]> wrote:



Hi, Joash,

thank you for your email. I was very relieved to hear that someone was
already doing this.

Can you please tell me more about your test? Why did you test it in a
first place?

For me, we need to come up with a system that needs to:

1. Handle 5,000 inbound SIP calls

2. offer IVR capability

3. Billing


You'd probably have to do some of your own work on this.  * makes 'CDR' 
records but...well...you have to be careful how you do your scripts if you 
want legible/useable CDRs.  There are some apps out there though that will 
process and do some sort of billing for CDRs not sure of what where.




I thought that Asterisk would be up to the task, but, I am not sure as
to:

1. How many servers should I consider? 4? 10? Obviously, we will be
talking about probably core Xeon servers if this is what we need.


I'd say atleast 10maybe more...depending wholly on codec/transcoding 
and amount of IVR scripting.




2. How hard would it be to implement?


Well...since your not well versed with *, and you're having trouble 
understanding the difference between a protocol and a codec, it might be 
really difficult for you.  You might want to farm it out.  There are a LOT 
of * consultancies out there now.  If you can get up to speed on asterisk 
pretty quickly and the various protocols and codecs then it's not 
impossible.  The kicker is all the management/maintenance UI's and such. 
But you might be able to use something like Signates sigMAN (never used it 
or their products).




3. How bad is g729 quality?

4. IVR : if the call is SIP, can we do prompts without transcoding?


You're confusing protocols with codec's here again.  SIP is not a codec. 
That said if your SIP client is using GSM and there are GSM prompts 
available then the asterisk playback functions will use the GSM encoded 
prompts.


Earlier you'd mentioned using POTS lines coming in/out.  If you're 
gatewaying 5k POTS lines you'll need a lot of machines.  Because you'll be 
doing  a lot of transcoding POTS is ulaw or alaw (depending on where in the 
world you are) and unless you use (uncompressed) ulaw or alaw on your SIP 
clients (very unlikely scenario) you'll be transcoding to/from GSM. G.729, 
or whatever you're using.




Any other suggestions that you might have would really be appreciated.





 Joash Herbrink <[EMAIL PROTECTED]> wrote:



I have tested an asterisk server with over 5000 concurrent calls.

The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet
connection on a cisco 3560 switch.



This works, but puts some serious stresses on the system.

Why don't u considered using g.729 codec, this will at least lower the
bandwidth consumption significantly, and, you can overcome the CPU
resource issue by just using a server grade multi CPU xeon server.



I would never the less still connect the system via 2 ethernet
connections, just for some redundancy, as mentioned before in this
thread.



Bandwidth should be about 24 kbps (half duplex) per call



So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just
fine.



Joash



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dustin
Wildes
Sent: Wednesday, February 01, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
question



Dinesh Nair wrote:












On 02/01/06 09:29 Damon Estep said the following:







Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -



full duplex.







Have you ever seen a NIC or switch that can run GigE full duplex at 80%



utilization and not at least start to fall apart?











additionally, 5000 simultaneous SIP calls at 20ms intervals will send,







5,000 * 50 * 2 = 500,000 packets per second (full duplex).







not too many boxes can handle such packet load, in spite of the



relatively small packet sizes.








Why not bond multiple NICs together to do a load balance output?  Would

provide redundancy as well.



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--
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into trouble of all kinds."
-- Samuel Butler
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[Asterisk-Users] delaying "answer" for a number of rings or an amount of time

[Brian J. Murrell]

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap
channel) by some period of time, either a number of rings or just a
number of seconds.

I have tried this:

[from-pots]
exten => s,1,Wait(30)
exten => s,n,Answer
...
exten => s,n,Dial(SIP/brian&SIP/joe,10,H)
exten => s,n,Voicemail(u2001)
exten => s,n,Hangup
exten => s,103,Voicemail(u2001)
exten => s,104,Hangup
exten => h,1,Hangup
exten => i,1,Hangup

but that doesn't work.  It seems that as soon as the first or second
ring is detected, Asterisk has decided it will answer the line, it just
waits 30 seconds to do it.

The problem I have is that the POTS line that Asterisk is on is shared
by handsets, and if somebody picks up a handset soon enough, I don't
want Asterisk to pick up the line.  Yeah, Asterisk as an answering
machine... until I can get at least one FXS interface, anyway.

Any ideas?

b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell


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[Asterisk-Users] How to handle "provider UNREACHABLE" in the dialplan?

[Ronald Wiplinger]

How to handle this in the dialplan?

voipbuster/   194.221.62.201  5060 UNREACHABLE
voipstunt/x 194.120.0.200   5060 UNREACHABLE

a reload shows than:

voipbuster/   80.239.235.200 5060 UNREACHABLE
voipstunt/x   194.120.0.200   5060 UNREACHABLE

bye

Ronald Wiplinger
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[Asterisk-Users] 5,000 concurrent calls system rollout question

[Vic]

Hi,
several of your mentioned signant as a viable option.
Has anyone ever used them? Are there any reviews for their products?
Did they just put together a lot of Asterisks into a large scale PC? (I am still struggling with the concept)
Thanks,
Vic

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RE: [Asterisk-Users] 5,000 concurrent calls system rollout question

[Vic]

Hi, Joash,
thank you for your email. I was very relieved to hear that someone was already doing this.
Can you please tell me more about your test? Why did you test it in a first place?
For me, we need to come up with a system that needs to:
1. Handle 5,000 inbound SIP calls
2. offer IVR capability
3. Billing
I thought that Asterisk would be up to the task, but, I am not sure as to:
1. How many servers should I consider? 4? 10? Obviously, we will be talking about probably core Xeon servers if this is what we need.
2. How hard would it be to implement?
3. How bad is g729 quality? 
4. IVR : if the call is SIP, can we do prompts without transcoding? 
Any other suggestions that you might have would really be appreciated.
 
 
 Joash Herbrink <[EMAIL PROTECTED]> wrote:







I have tested an asterisk server with over 5000 concurrent calls.
The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet connection on a cisco 3560 switch.
 
This works, but puts some serious stresses on the system.
Why don't u considered using g.729 codec, this will at least lower the bandwidth consumption significantly, and, you can overcome the CPU resource issue by just using a server grade multi CPU xeon server.
 
I would never the less still connect the system via 2 ethernet connections, just for some redundancy, as mentioned before in this thread.
 
Bandwidth should be about 24 kbps (half duplex) per call
 
So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just fine.
 
Joash
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wildes
Sent: Wednesday, February 01, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
 
Dinesh Nair wrote:
 
> 
> 
> On 02/01/06 09:29 Damon Estep said the following:
> 
>> Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps -
>> full duplex.
>> 
>> Have you ever seen a NIC or switch that can run GigE full duplex at 80%
>> utilization and not at least start to fall apart?
> 
> 
> additionally, 5000 simultaneous SIP calls at 20ms intervals will send,
> 
> 5,000 * 50 * 2 = 500,000 packets per second (full duplex).
> 
> not too many boxes can handle such packet load, in spite of the 
> relatively small packet sizes.
> 
 
Why not bond multiple NICs together to do a load balance output?  Would 
provide redundancy as well.
 
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Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Armin Schindler]

On Thu, 2 Feb 2006, Bartosz Jozwiak wrote:
> Dear all,
> 
> I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and run
> with chan_capi.
> Is anybody using that card ? Would appreciate any feedback.

I have the non-V version of that card multiple times in use with
perfect results.
Do you need specific information?

Armin

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[Asterisk-Users] Events when the target answer

[Ezequiel A. Sculli]

 

Hi Group, I am developing a application, this use
“Manager API” to connect with Asterisk. But when I call to an
external number (over a zap channel), I don’t receive any event when the
target answer, Who can help me?, Which event notify me that the phone call was
answered?

 Thank you. 

 

Ezequiel






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Re: [Asterisk-Users] Default value for ASTERISK_VERSION_NUM

[Kevin P. Fleming]

Leo Ann Boon wrote:


/*
* version.h
* Automatically generated
*/
#define ASTERISK_VERSION "1.2.4"
#define ASTERISK_VERSION_NUM 00


This was a bug in the Makefile; it has been corrected in Subversion and 
will part of the 1.2.5 release. Sorry for the inconvenience.

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[Asterisk-Users] Sip - no peer or user found on incoming call

[Administrator TOOTAI]

Hi list,

I try to connect to a GW which have one domain eg sip.mydomain.com and 
have few IPs related to this domain. I register * to this domain with 
host=sip.mydomain.com and type=user. So DNS will decide on which IP of 
my domain I will register (or redirection on the GW side).


If an incoming call arrive, I would guess that, as type=user, it will 
not try to match the IP from INVITE as I want to match on username. But 
this is not true, I always have in logs "Found no matching peer or user 
for 'xxx.xxx.xxx.xxx:5060'" and asterisk then try to find a  
extension in the SIP default context. I tried to play with deny/permit 
without luck.


The call is finishing properly _only_ when the IP which with my * is 
registred to the GW match this from the incoming call, and then doesn't 
matter if type=user or type=peer, which is normal according to 
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer.


I'm running Asterisk SVN-trunk-r8643M built by root @ keewi on a i686 
running Linux on 2006-01-25 14:50:51 UTC


Here is relevant part of my sip.conf

register => :@sip.mydomain.com/

[IN-UserName]
type=user
username=
fromuser=
fromdomain=
secret=
context=incoming-GW
;deny=0.0.0.0/0.0.0.0
;permit=xxx.xxx.xxx.xx0/32
;permit=xxx.xxx.xxx.xx1/32
;permit=xxx.xxx.xxx.xx2/32
;permit=xxx.xxx.xxx.xx3/32
;permit=xxx.xxx.xxx.xx4/32
;permit=xxx.xxx.xxx.xx5/32
host=sip.mydomain.com
;insecure=invite,port   ;very
;nat=yes
;canreinvite=no
;qualify=1000
disallow=all
allow=g726

Thanks for any clue.

--
Daniel
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[Asterisk-Users] ISDN Eicon Diva Server V-BRI

[Bartosz Jozwiak]

Dear all,

I'm planning to buying Eicon Diva Server V-BRI for my 
asterisk server and run with chan_capi.

Is anybody using that card ? Would appreciate any feedback.

Bartosz
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RE: [Asterisk-Users] Directed Call Pickup

[Mimmus]

Same problem for me. Direct call pickup doesn't work. Global pickup is OK.
This is 'show features' output:

> show features 
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8 
Blind Transfer#   #1 
Attended Transfer #2 
One Touch Monitor *1 
Disconnect Call   *   *  

Dynamic Feature   Default Current
---   --- ---
(none)

Call parking

Parking extension   :   700
Parking context :   parkedcalls
Parked call extensions: 701-720 


'show modules' says that app_directed_pickup.so is loaded:

 app_directed_pickup.so Directed Call Pickup Application 0 

Then I have also:

show application Pickup
asterisk1*CLI> 
  -= Info about application 'Pickup' =- 

[Synopsis]
Directed Call Pickup

[Description]
  Pickup([EMAIL PROTECTED]): This application can pickup any ringing
channel
that is calling the specified extension. If no context is specified, the
current
context will be used.


Any help?

Thanks
Mimmus

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Re: [Asterisk-Users] meetme and dtmf

[Kevin P. Fleming]

Accursio Avona wrote:

The IVR records the conversation between the other partecipant to the 
conference and wait '#' to stop recording and a '1'  to save the file.


Then I really don't understand at all... this is not functionality that 
I would call an 'IVR'.


Can you show us the portions of the Asterisk dialplans that are involved 
here?

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[Asterisk-Users] Asterisk at SCALE 4x

[Ilan Rabinovitch]

Hello,

Asterisk will have strong presence at SCALE 4x, the 2006 Southern
California Linux Expo next week.
On the exhibit hall floor both Digium and SwitchVox will have booths
demonstrating asterisk and related products.

The event will be held on Feb 11th and 12th at the Los Angeles Airport
Radisson.  In addition to Asterisk focused sponsors, we will have 3
talks on the topic of Asterisk and open-source VoIP:

* Mark Spencer (Digium) - IP Communication: Open for Business
* David Mandelstam (Sangoma) - It's a whole new world -- open source
at the PBX, ready for prime time
* Tim Fritchel - Case Study Switching from Motorola to Asterisk

Other speakers include:  Hans Reiser, Chris Dibona, Andi Gutmans and
more.. For further details see the conference website at:
http://www.socallinuxexpo.org

Those interested in attending the show can use the promo code "AST06" 
to get 40% off full access passes. 
(http://www.socallinuxexpo.org/order/)
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[Asterisk-Users] RE: Rewind MusicOnHold?

[Brent Torrenga]

The native MOH type will, unless set to random=yes, play the music files in
the same order as they appear with an ls of the directory. (someone, anyone,
back me up here?)

I would place the greeting in the same MOH class as your actual music, and
name the file of the greeting something "less" than the filename of the
music file. Additionally, to avoid repeating the greeting, should the music
file play all the way through before an answer, you may want to make
additional copies of the music file, named something "greater" than the
greeting file.

Just a thought, never tried to do it myself.


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com

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Re: [Asterisk-Users] Call completes but then drops?

[Kevin P. Fleming]

Matt wrote:


I guess I'm more trying to figure out what
Feb  1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from
channel: SIP/102-9fda
means.


Pretty much what it says... the SIP endpoint dropped its end of the call 
and the Asterisk channel was hung up as a result.


Given the sheer number of bugfixes that have been made since you got 
that code, I would suggest that you are wasting time trying to debug 
this without upgrading :-)

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RE: [Asterisk-Users] POTS lines vs. using T1 to connectphoneservices?? HELP

[Michael Collins]

Kevin,

Are you in the US?  If so then you've probably got several carriers to
choose from.  In my experience analog lines have a flat expense of
$20-$25 per month.  That equates to about $140-$175 per month in flat
fees, plus you have usage on top of that. (Your experience may vary.) I
am currently experimenting with a company out of NY called Digizip
(www.digizip.com) that sold me a Qwest PRI for about $150/month flat fee
plus usage in the neighborhood of $.015 per minute. (Month-to-month
term, no contract!) 

A PRI like this is attractive because you have the capability of having
23 simultaneous conversations, plus you can do DID.  One drawback is the
inability to do a "Centrex transfer" (aka DID to DOD transfer or "off
net transfer") but that usually isn't a big deal.

One other note: in the US it is considered a legal requirement to have a
CSU on any T1 circuit.  However, it is not technically necessary.  Also,
some terminating equipment has the "CSU" built right in - e.g. Cisco
T1/CSU WIC for their routers.  I'm running 12 different T1 circuits,
each with a CSU.  I like having the CSU for testing and monitoring line
conditions.

If you go with a PRI (or any other T1-style circuit) then it's just
matter of getting the right card for your system.  The Digium and
Sangoma cards have fine reputations for use in production machines.  The
advanced (and more expensive) models have echo cancellation built in.
I'm currently using a knock-off of the older Tormenta2 Zapata card but
that's only for testing.  In a production environment I'll mostly likely
upgrade to a better card.

HTH and sorry for the ramble!

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Steil
Sent: Thursday, February 02, 2006 7:46 AM
To: Asterisk User List
Subject: RE: [Asterisk-Users] POTS lines vs. using T1 to
connectphoneservices?? HELP

Thanks...just need to see what the cost is...compared to getting 6
lines..

-Original Message-
From: Tom Vile [mailto:[EMAIL PROTECTED] 
Sent: Thursday, February 02, 2006 9:58 AM
To: Asterisk User List
Subject: Re: [Asterisk-Users] POTS lines vs. using T1 to connect
phoneservices?? HELP

A fractional T1 is what I would suggest and it is easy to setup and
configure.  You should only need to plug in the T1 line directly into
the T1 Card on the server.  The provider will supply the equipment to
terminate the line on your premises.

On 2/2/06, Kevin Steil <[EMAIL PROTECTED]> wrote:
> Need help...I need to install a card to terminate 7 lines...I
> have not order the phone lines yet...I can either do analog lines 1FBs
> or order a fractional T1...any suggestions on what hardware would be
> easier to install and configure...also if I went with a T1...do I need
> an external CSU/DSU or anything or does it just plug into the T1
> card...thanks..
>
> Kevin J. Steil
> Steil Technologies
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856

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Re: [Asterisk-Users] Rewind MusicOnHold?

[Dan Journo]

1) Answer incoming call 
 
exten => s,1,Answer() 
2) Begin dialing an extension 
 3 ) While extension is ringing play a welcome message to the caller
 
Here you got a problem. What do you do if callee picks up too fast.
In my situation, the caller wont pickup too fast. The message is 10 seconds long, and the shortest time for the person to answer is around 20 seconds. It will never be less.
 
Thanks
Dan 
On 02/02/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:

 



From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Rewind MusicOnHold? 

I thought someone was going to say that.
Does anyone know a way to do the following:-
 
1) Answer incoming call 
 
exten => s,1,Answer() 
2) Begin dialing an extension 
 3 ) While extension is ringing play a welcome message to the caller
 
Here you got a problem. What do you do if callee picks up too fast.
 
so I would
 
exten => s,2.PlayBack(mesage-to-caller0
exten => s,3,Dial(SIP/123||m)
 
 
 4) Then play MoH until the extension is answered
5) Connect the incoming and outgoing when the extension is answered.
 
Thanks
Dan 
On 02/02/06, Alexander Lopez <
[EMAIL PROTECTED]> wrote: 

Turn phone over and shake!
:-)
 
MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF, or RW.
 
(gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended)
  
 



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Dan Journo
Sent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? 




Does anyone know how to rewind the music on hold?
 
Thanks
Dan Journo___--Bandwidth and Colocation provided by 
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RE: [Asterisk-Users] Asterisk on laptop connected to POTS line

[Mr. James W. Laferriere]

Hello All ,

On Thu, 2 Feb 2006, [EMAIL PROTECTED] wrote:

 Original Message 
Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS line
From: Tzafrir Cohen <[EMAIL PROTECTED]>
Date: Thu, February 02, 2006 9:15 am
To: asterisk-users@lists.digium.com

On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote:


Anyone know of any equipment that I can use to connect a
laptop running asterisk to a POTS line (RJ11) ?


Look at Xorcom's USB channel Bank.


Which is a great product and you should all get one (and the fact that
I'm a Xorcom employee has nothing to do with this recommendation), but
sadly, still lacks FXO ports.


If Xorcom could offer something similar with 2-4 FXOs I'd just have to
buy at least one. Heck of an idea for a product, a quad FXO adapter
interfaced to Asterisk via local USB port. Wow!

If one could get this in 1-3 FXO & 1-3FXS ports(*) in an
apropriate combination ...  Where the USER can select which
combo s/he wants at home ,  Not by buying a hardwired device .
Then that would be something to buy .

(*) 1FXO 3FXS ,  2FXO 2FXS ,  3FXO 1FXS .
My $.02 ,  JimL
--
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
|  http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr   |
+--+
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Re: [Asterisk-Users] Rewind MusicOnHold?

[Dan Journo]

The reason we want to do it this way, is that we'd like to start dialing at the beginning, so that when the message finishes playing, the caller has actually already waited 10 seconds, leaving 10 seconds before the call is answered.

 
Thanks
Dan 
On 02/02/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:

 



From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Rewind MusicOnHold? 

I thought someone was going to say that.
Does anyone know a way to do the following:-
 
1) Answer incoming call 
 
exten => s,1,Answer() 
2) Begin dialing an extension 
 3 ) While extension is ringing play a welcome message to the caller
 
Here you got a problem. What do you do if callee picks up too fast.
 
so I would
 
exten => s,2.PlayBack(mesage-to-caller0
exten => s,3,Dial(SIP/123||m)
 
 
 4) Then play MoH until the extension is answered
5) Connect the incoming and outgoing when the extension is answered.
 
Thanks
Dan 
On 02/02/06, Alexander Lopez <
[EMAIL PROTECTED]> wrote: 

Turn phone over and shake!
:-)
 
MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF, or RW.
 
(gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended)
  
 



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Dan Journo
Sent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? 




Does anyone know how to rewind the music on hold?
 
Thanks
Dan Journo___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Directed Call Pickup

[Bob Goddard]

On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote:
> Hi All
>
> I am having problems with Directed Call Pickup in Asterisk 1.2.1
>
> If extension 100 is ringing, a user at another extension is supposed to
> be able to dial *8100 and pickup the call to 100.  It isn't working for
> me and I cannot figure out why.
>
> I have in features.conf:
>
> pickupexten = *8

At the CLI, "show features" should tell you if it is configured.
If so, you need to tell us what happens on the console.
If not, then you are liable to get asked "my car does not work,
does anyone know why?".


B

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RE: [Asterisk-Users] Rewind MusicOnHold?

[Alexander Lopez]

 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dan 
  JournoSent: Thursday, February 02, 2006 11:18 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Rewind MusicOnHold?
  
  I thought someone was going to say that.
  Does anyone know a way to do the following:-
   
  1) Answer incoming call 
   
  exten => s,1,Answer() 
  2) Begin dialing an extension 
   3 ) While extension is ringing play a welcome 
  message to the caller
   
  Here 
  you got a problem. What do you do if callee picks up too 
  fast.
   
  so I 
  would
   
  exten => s,2.PlayBack(mesage-to-caller0
  exten => s,3,Dial(SIP/123||m)
   
   
   4) Then play MoH until the 
  extension is answered
  5) Connect the incoming and outgoing when the extension is 
answered.
   
  Thanks
  Dan 
  On 02/02/06, Alexander 
  Lopez <[EMAIL PROTECTED]> wrote: 
  
Turn 
phone over and shake!
:-)
 
MOH 
plays in a loop, It has no player controls. You can't Pause, Stop, 
FF, or RW.
 
(gasp!) 
You can try killing the mpg123 or whatever process you use for MOH. Asterisk 
should restart it if it needs to. (high-unrecomended) 
 
 



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
Dan JournoSent: Thursday, February 02, 2006 10:27 
AMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: [Asterisk-Users] Rewind 
MusicOnHold? 


  
  Does anyone know how to rewind the music on hold?
   
  Thanks
  Dan 
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Re: [Asterisk-Users] Rewind MusicOnHold?

[Garth van Sittert]

Hi Dan

Have a look at setting up queues.

Kind Regards
Garth


Dan Journo wrote:

I thought someone was going to say that.
Does anyone know a way to do the following:-
 
1) Answer incoming call

2) Begin dialing an extension
3) While extension is ringing play a welcome message to the caller
4) Then play MoH until the extension is answered
5) Connect the incoming and outgoing when the extension is answered.
 
Thanks

Dan

 
On 02/02/06, *Alexander Lopez* <[EMAIL PROTECTED] 
> wrote:


Turn phone over and shake!
:-)
 
MOH plays in a loop, It has no player controls. You can't Pause,

Stop, FF, or RW.
 
(gasp!) You can try killing the mpg123 or whatever process you use

for MOH. Asterisk should restart it if it needs to.
(high-unrecomended)  
 


*From:* [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
] *On Behalf Of
*Dan Journo
*Sent:* Thursday, February 02, 2006 10:27 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] Rewind MusicOnHold?

 


Does anyone know how to rewind the music on hold?
 
Thanks

Dan Journo


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--
Garth van Sittert
BSc (Physics & Computer Science)
-
Mobile: +27 (0)83 791 6662
Email:  [EMAIL PROTECTED]
Phone:  08600 BITCO
Web:www.bitco.co.za 


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RE: [Asterisk-Users] Callerid Name

[Alexander Lopez]

Look at
 http://bugs.digium.com/view.php?id=1192

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> John Bittner
> Sent: Thursday, February 02, 2006 11:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Callerid Name
> 
> Anyone know why zaptel would ignore a facility message from 
> an ISDN PRI.
> I am trying to get Callerid name to work. The carrier says it 
> on and I see it in the pri debug but asterisk never gets it.
> 
> Any help would be appreciated.
> 
> Thanks
> 
> John Bittner
> Simlab.net
> 
> 
> 
>  Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 572/0x23C) (Terminator) Message type: 
> > ALERTING (1) [1e 02 81 88]I> Progress Indicator (len= 4) [ Ext: 1  
> > Coding: CCITT (ITU) standard (0) 0:
> 0   Location: Private network serving the local user (1)
> >   Ext: 1  Progress Description: Inband
> information or appropriate pattern now available. (8) ]
> -- SIP/69.60.198.130-5119 is ringing < Protocol 
> Discriminator: Q.931 (8)  len=36 < Call Ref: len= 2 
> (reference 572/0x23C) (Originator) < Message type: FACILITY 
> (98) < [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 42 55 
> 4c 4b 41 4e 27 53 2c 48 45 41 4c 54 48] < Facility (len=31, 
> codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 
> 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'BULKAN', 0x27, 'S', 
> 0x2c, 'HEALTH' ]
> -- Processing IE 28 (cs0, Facility)
> Handle Q.932 ROSE Invoke component
> 
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[Asterisk-Users] Directed Call Pickup

[Garth van Sittert]

Hi All

I am having problems with Directed Call Pickup in Asterisk 1.2.1

If extension 100 is ringing, a user at another extension is supposed to 
be able to dial *8100 and pickup the call to 100.  It isn't working for 
me and I cannot figure out why.


I have in features.conf:

pickupexten = *8

Kind Regards
Garth


--
Garth van Sittert
BSc (Physics & Computer Science)
-
Mobile: +27 (0)83 791 6662
Email:  [EMAIL PROTECTED]
Phone:  08600 BITCO
Web:www.bitco.co.za 


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[Asterisk-Users] Re: Euro-ISDN

[Aldo Bergamini]

[EMAIL PROTECTED] is believed to have said: 

>With BRIstuff you get to use ztcfg, etc.
>
>Cannot say anything about mISDN, CAPI...

Francesco,

thank you; this is important to know

Aldo

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[Asterisk-Users] Re: Euro-ISDN

[Aldo Bergamini]

[EMAIL PROTECTED] is believed to have said: 

[setup tool]
>Sorry, I cannot answer that one. I don't know enough about these cards and 
>their drivers.

Armin,

thanks alot.  One has to do some research and experimentation on his own
every now and then; and see if there is anything interesting that might
even end up in the wiki...

;-)

>> So in the end there a lot of reasons to go for a 'better' card.
>
>Yes, a lot reasons. But actually, it depends on what you need and what you
>want to do.

You are right..

I was asking about ISDN cards to see if there was any simple integration
path for some more expensive units (where one might use those with ISDN
interfaces first).

But now I'll have to fiddle with the newly arrived GSM-gateway first. 
And this is a nice, little analog unit.

Thank you very much,
Aldo

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