date:20050802

On Tuesday 02 August 2005 06:35, Obelix wrote:
 I have seen the Asterisk info about TDMoE - does this mean that the
 Asterisk card will modulate the signal on the Ethernet cable to allow it
 plug directly into a proper TDM connection?

TDMoE is just a method of taking the 8000Hz, 8-bit ulaw/alaw audio from a TDM 
interface and stuffing them into raw ethernet frames.  The destination MAC 
must obviously be on the same ethernet network.  Different segments are fine 
since any competent switch will be able to get the frames to their 
destination.

In a nutshell.  Take a raw ethernet frame.  You have the ethernet header and 
the payload.  Take the destination MAC and put it in the header, and take the 
TDM data and put it in the payload.  There may be a little more in the 
payload (timestamps I would suspect).  Send it out to the card.

That's it.

-A.
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Re: [Asterisk-Users] Minimum CPU required for 60 calls

On Tuesday 02 August 2005 06:16, Obelix wrote:
 I am interested in how much CPU and RAM asterisk requires for call
 handling.

I *really* dislike these kinds of questions.

Grab some hardware and try it.  It is the *ONLY* way you will know for sure.  
Grab a single processor Pentium 4 or Celeron system and do some testing.  I'm 
sure you have one sitting around somewhere you can use for a test, even if 
you have to put a different hard drive in it for the test.

Typically speaking, if you have to ask these kinds of questions you are 
NOWHERE near the level of competence in Asterisk to try and skimp and save on 
the hardware.  That is not meant as an insult, either.  Get it working, THEN 
start looking to pinch the pennies.  You will only be disappointed otherwise.

-A.
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Re: [Asterisk-Users] Astcc Configuration Problem

2005-08-02 Thread Darren Wiebe


Try this line in /var/lib/astcc and see if it helps.

chown apache * and

chown -R apache * from /var/www/cgi-bin.

If it does not help, do an ls -l in /var/www and see who owns it.  
Then repeat the command above with the correct user. 


Darren Wiebe
[EMAIL PROTECTED]

chawki hammoud wrote:


Hi:

How do I check it? 


Thanks

--- Darren Wiebe [EMAIL PROTECTED] wrote:

 


Check and make sure that astcc-config.conf is owned
by the same process 
that owns apache.  Usually the problem is that
astcc-admin cannot write 
to the file due to permission problems.


Darren Wiebe
[EMAIL PROTECTED]

chawki hammoud wrote:

   


Hi:

I used astcc to create database. After I get the
message database created, I save the configuration
 


and
   


I move to the next step to assign trunk and route.
 


But
   


I get the message:
Database unavailable -- please check configuration
Cannot edit routes until database is configured

I checked the databse and it's in mysql, the file
/var/lib/astcc/astcc-config.conf is empty.
astcc-admin.cgi is supposed to write a file based
 


on
   


the configuration, but for some reason doesn't
I inserted data manually into the databse tables
 


and
   


astcc works fine on asterisk and write data into
 


the
   


database.
Why the web browser can't see the database it
 


created?
   


could apache be the problem although it's running?

Regards;
Chawki



 



   


Start your day with Yahoo! - make it your home page
 

http://www.yahoo.com/r/hs 


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Re: [Asterisk-Users] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx

2005-08-02 Thread Adam Dobrin

You aren't dealing with analog phones, and you aren't transmitting DTMF 
signals.. the functional difference between analog and digital systems 
kindof precludes what you are looking to do.. meanwhile, once the entire 
number has been dialed, the outgoing call should be started almost 
instantaneously..


maybe set the initial context so that the longest dial string is the 
length of the extension..?


Frank Sautter wrote:


Maik Schmitt schrieb:


one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
  telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx it 
takes

about 3 seconds before the asterisk start to dial.
where does this delay come from?
has it to do with 'overlapdial=yes'?



This is normal behaviour if you use '.' in your extensions.conf. Use 
'!' instead and Asterisk will start dialing immediately.


when i change '.' to '!' then the overlap digits get lost. this means 
the longest number dialled on my telco line is as long as there are 
abigous matches in the dialplan.
isn't there a way to start dialling after one received enough digits 
to decide which path to dial and then still transmit the remaining 
(overlapping) digits?


regards
 frank

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[Asterisk-Users] SIP Debug

2005-08-02 Thread Michael Anuzis

Using Asterisk Management Portal with Broadvoice. It used to work just
fine; calls would come in and be answered with no trouble at all. A
few weeks ago with no configuration changes at all Asterisk stopped
picking up calls and started giving a busy signal whenever someone
calls.I've tried rebooting the system many times, and sip show
registry shows it's registering correctly with Broadvoice. Sip debug
shows the UDP packets correctly hit the system on port 5060, but the
call is rejected\busy instead of answered.

Here's a SIP debug or a call coming in and being busy. Any clues? 

Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
From: sip:[EMAIL PROTECTED];user=phone;tag=xz13
To: sip:[EMAIL PROTECTED];user=phone
Via: SIP/2.0/UDP 147.135.12.128:5060
Contact: sip:[EMAIL PROTECTED]:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID: sip:[EMAIL PROTECTED];screen=yes;party=calling;privacy=off
Content-Length:  273
Content-Type: application/sdp

v=0
o=2475101431 10 10 IN IP4 147.135.12.247
s=-
c=IN IP4 147.135.12.250
t=0 0
m=audio 18092 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

12 headers, 12 lines
Using latest request as basis request
Sending to 147.135.12.128 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.12.250:18092
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Found peer 'sip.broadvoice.com'
Looking for  in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.12.128:5060
From: sip:[EMAIL PROTECTED];user=phone;tag=xz13
To: sip:[EMAIL PROTECTED];user=phone;tag=as54c1e248
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


 to 147.135.12.128:5060
asterisk1*CLI
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Re: [Asterisk-Users] Minimum CPU required for 60 calls

2005-08-02 Thread Adam Dobrin

And as much as you dislike these kinds of questions; its unfortunate 
that the community doesn't have any good answers to them available--they 
should be.  It would be great if we could get some independent 
verification of digium's claims/figures.


voip-info:
http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office+100



Andrew Kohlsmith wrote:


On Tuesday 02 August 2005 06:16, Obelix wrote:
 


I am interested in how much CPU and RAM asterisk requires for call
handling.
   



I *really* dislike these kinds of questions.

Grab some hardware and try it.  It is the *ONLY* way you will know for sure.  
Grab a single processor Pentium 4 or Celeron system and do some testing.  I'm 
sure you have one sitting around somewhere you can use for a test, even if 
you have to put a different hard drive in it for the test.


Typically speaking, if you have to ask these kinds of questions you are 
NOWHERE near the level of competence in Asterisk to try and skimp and save on 
the hardware.  That is not meant as an insult, either.  Get it working, THEN 
start looking to pinch the pennies.  You will only be disappointed otherwise.


-A.
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[Asterisk-Users] How to create a secret code to use my [EMAIL PROTECTED] server's long distance plan from a public phone

2005-08-02 Thread Adrien Laurent

Hello everyone,


I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.

When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.

How can I setup a secret password in the extension.conf, so that my
asterisk server can allow me to make long distance call ?


Thanks a lot,

Adrien
-- 
Adrien Laurent
[EMAIL PROTECTED]
www.modulis.ca
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Re: [Asterisk-Users] Re: two UA with the same usr/pwd

Many people seem to want this feature.  I think they are just 
confused.  I've never actually heard of a good reason to let multiple 
devices register with the same username/secret.  Most of the time they 
 want a call to ring on multiple devices and they are trying to make 
a device == extension, which is not correct.  A device is a device and 
an extension is an extension and they are not the same thing and there 
is no 1-to-1 mapping between them.


Victor Alvarez wrote:

 I really think this matter deserves attention. I have been asked many times 
about it.

 Regards,
  Victor. 




Hello,

I can understand why asterisk is designed to not to allow two UAs with the same 
usr/pwd, 
http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but 
I have to find a solution for this.

My first option is use SER as an extension end of Asterisk, to allow more than 
one SIP endpoint to register with the same details 
http://www.voip-info.org/wiki-Asterisk+at+large. I wonder if there is another 
way to do this. Of course, I am talking about a SIP proxy behaviour, 
simultaneous registration, both phones ringing at the same time and first to 
answer gets the call.

Kind regards,
Victor.




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[Asterisk-Users] Polycom SoundPoint 600 : 10 seconds of delay when answering a call.

2005-08-02 Thread Ken Dresdell









Hello everyone, I have just received 3
brand new Polycom SoundPoint
IP 600 from voipsupply.com and I have the exact same problem on
all of them. When I receive a call, the phone is ringing correctly but when I
answer it, it takes exactly 10 seconds before I can hear the caller. I also
have SoundPoint 300 and 301 and I dont have
that problem with those. I'm using Asterisk 1.0.7.



I checked the user guide and admin guide
from Polycom but didnt see anything
interesting.



Does anyone encounter that problem? Any idea?



Ken






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RE: [Asterisk-Users] Re: IAX Devices Recommendation

You may have bought the Chinese Versions and hence the problem in slow
response.

Have you tried the US versions available from http://www.iareaphone.com
?

-S


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Redstone
Sent: Tuesday, August 02, 2005 12:32 AM
To: Asterisk User
Subject: [Asterisk-Users] Re: IAX Devices Recommendation

Hi

We purchased the AT320-EE IAXtalk phone from www.iaxtalk.com which
ocnnects to our own asterisk server.

Good value, a little tricky to set up - the instructions they supply to
which they give you a link on their web site are OK, but their are some
gaps which the asterisk wiki pages fill well - cannot find this at the
moment but it explains how to do resets.

IN summary you buy the phone and then upload the firmware for IAX2
protocol. 
Configuration is via web browser which works well. Automaticlaly logs
in.

Works well. Slightly slower to respond than (say)  firefly softphone
which we use for most users - the hardphone is for reception and as
backup in case of computer failure.

Paul Redstone
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 7

2005-08-02 Thread Nguyen Trung Tin

Hello ALL

SS7 for asterisk release http://www.footnotess7.com/. but i not yet account to download.
any body have SS7. could you like send to me.

thanks
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[Asterisk-Users] IAX Devices Recommendation

Graham,

Digium IAX2 FXS unit called IAXY is just no good. I would say that it is
garbage. 

Try the IAX2 ATA ( AG168 sold as Netweb ATA-100) with a life line port
made by Atcom and available from http://www.iareaphone.com

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham
Pearson
Sent: Monday, August 01, 2005 3:07 PM
To: Asterisk Users
Subject: [Asterisk-Users] IAX Devices Recommendation

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS Adapter but
unable to find a Telephone that supports the IAX Protocol. Any
Recommendations or is the Digium FXS Adapter the way to go.


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[Asterisk-Users] IAX Devices Recommendation

With due respect to Digium and Mark Spencer and the greatest protocol he
defined, I have used IAXY and I regret to say that IAXY at $99 is plain
garbage compared to the $49 ATA made by ATCOM. 

Try the ATCOM AG168  sold as ATA-100 by iareaphone.com. This has an
additional lifeline port and gives the best bang for the buck.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham
Pearson
Sent: Monday, August 01, 2005 3:07 PM
To: Asterisk Users
Subject: [Asterisk-Users] IAX Devices Recommendation

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS Adapter but
unable to find a Telephone that supports the IAX Protocol. Any
Recommendations or is the Digium FXS Adapter the way to go.


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RE: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] server's long distance plan from a public phone

2005-08-02 Thread Giles Coochey

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Adrien Laurent
 Sent: 02 August 2005 14:56
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] How to create a secret code to use 
 [EMAIL PROTECTED] server's long distance plan from a public phone

 Hello everyone,

 I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
 I have a trunk connected to a voip provide, asteriskout.

 When I call my server from a public phone, I want to route this call
 to the asteriskOUT trunk so that I can make long distance calls.

 How can I setup a secret password in the extension.conf, so that my
 asterisk server can allow me to make long distance call ?

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

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message and all copies and backups thereof.

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RE: [Asterisk-Users] Asterisk PSTN connectivity





Use Googleextensively andthe WIKIsitehere http://www.voip-info.org/wiki-Asterisk, 
till you become familiar with the architecture of Asterisk. probably for a 
couple of months. 

You can come back here if you still have any questions 
at that time and all the member here would be happy to answer all your 
questions.

Seshu



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Nil 
SSent: Tuesday, August 02, 2005 5:07 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
PSTN connectivity

Hello Everybody,

I am a new user in this group.
I have installed asterisk on my test linux machine and setup the call 
from one asterisk user to another asterisk user successfully. It is working 
great.

Now i want to setup the call from one asterisk user to any PSTN user in the 
world or vice versa. Could you please help me out in this?
Please guide me how to do this as I am completely unaware of this 
Asteris PSTN connectivity.
Please suggest me some configuration steps also.

Waiting for positive reply.

Thanks.Nil





Do you Yahoo!?Yahoo! 
Mail - You care about security. So do we.



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Re: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] server's long distance plan from a public phone

2005-08-02 Thread Ariel Batista


Adrien Laurent wrote:

Hello everyone,


I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.

When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.


Your going to have to add an extenion in the extensions_custom.conf for a 
DISA setup. Depending on what your extensions look like it's fairly easy to 
do.  Your system should be answered by the Digital Reception for this to 
work correctly or a dedicated DID routed to it. This is a quick sample.


lets give an extension like 300.

exten = 300,1,Goto(custom-disa,s,1)

[custom-disa]

exten = s,1,Authenticate(1234) ; any number you want for a password here.
exten = s,2,DISA,no-password|from-internal
exten = s,3,Hangup


How can I setup a secret password in the extension.conf, so that my
asterisk server can allow me to make long distance call ?


Thanks a lot,

Adrien 

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[Asterisk-Users] Voicemail/Password Issue

2005-08-02 Thread Joseph

Does anyone know if the 3rd patch listed on this bug fixes the seg fault
problem related to voicemail?

http://bugs.digium.com/view.php?id=4800


-- 
respectfully, Joseph


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Re: [Asterisk-Users] call center 20 seats

2005-08-02 Thread Zeeshan


mattf wrote:

What kind of call center: inbound, outbound or both? 
 


It will be inbound 90%+ as I only need 2 seats for outgoing.

how many lines per agent will you have? 
 


one line per agent.


what kind of trunks will you be using?
 

Don't know yet. I am open for options and basically I don't want it to 
be bandwidth or process hungry.


do you need to tie into an existing database? 
 


No.


do you want screen-pops?

 


YES.

Let me know if you have any more questions.

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Re: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] server's long distance plan from a public phone

2005-08-02 Thread Doug Logan

Page 4 documents the Authenticate Feature. I'm a Newbie, so I can't give you 
much more help beyond that, but it should point you the right direction.

http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-pdf/vm1.pdf

Subject: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] 
server's long distance plan from a public phone
   From: Adrien Laurent [EMAIL PROTECTED]
   Date: Tue, 2 Aug 2005 09:55:32 -0400
 To: asterisk-users@lists.digium.com

Hello everyone,

I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.

When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.

How can I setup a secret password in the extension.conf, so that my
asterisk server can allow me to make long distance call ?

Thanks a lot,

Adrien
-- 
Adrien Laurent
[EMAIL PROTECTED]
www.modulis.ca
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Re: [Asterisk-Users] Queue/Agents

2005-08-02 Thread Waldo Rubinstein

If you don't mind, can you follow up with this on the list. I'm  
interested in learning how the different agent logins affect FOP.


Thanks,
Waldo

On Aug 2, 2005, at 1:46 AM, Nicolás Gudiño wrote:


Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything  
else but

no luck on the agent part..



How are your agents loging into queues? Depending on that you should
use slightly different configurations. Contact me off list if you need
assistance.

Regards,

--
Nicolás Gudiño
Buenos Aires - Argentina
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[Asterisk-Users] Asterisk as PSTN gateway, voice mail server with SIP

2005-08-02 Thread Innocent Evil

Hello,

I am sure this has been answered so many times as it is one of the most
fundamental features of Asterisk.

Here is my scenario,

I have setup my asterisk server with a TDM400p which have one FXO and FXS
card.
My SIP server is up and its working fine only in SIP network ( I used ser)
For my daily use telephone, I have a VoIP telephone from a major service
provider.

What I want is to hook my telephone line to Asterisk server FXO port.
So Asterisk is going to work as PSTN gateway for my SIP server.

How do I would do these:
1. If I receive a call to my telephone line, I would like to forward it from
my asterisk server to my SIP
phone.
2. Using my SIP phone, I would like to make PSTN call using my asterisk
server. How I know how to pass call to asterisk. But I dont know how to
receive that call from SIP server and initiate call using my  telephone line
on asterisk server.
3. Just in case, I am not able to answer from my sip phone, I would like
forward that call to asterisk so that caller can leave message.
4. I would like to retrive stored message from asterisk server.

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[Asterisk-Users] New release: Queue Statistics 0.1

2005-08-02 Thread Zoa



As promised, we just released the first version of the asteriskguru
Queue Statistics.

Screenshots and download at:
http://www.asteriskguru.com/tools/queue_stats.php

---
Small description:

The Asteriskguru queue statistics, is a PHP based program, which gives
anyone who uses queueing in Asterisk a deep insight in the quality of
the service which is delivered to their customers. It is fully
developped by the Asteriskguru developpers. The features we built into
it are:

   * complete overview of all incoming calls to your queue
   * complete overview of all taken and lost calls
   * graphical and table based representation of the quality of service
 provided to your clients
   * overviews of calltime and holdtime
   * selections can be made on
 o any chosen period of time
 o any queue
 o agents
 o date
 o hour
 o weekday
   * and it's free! (but not open)

Installation instructions will be added later on the day. (wrapping them
in html now).




signature.asc
Description: OpenPGP digital signature
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Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

2005-08-02 Thread Gonzalez Mata David

I'm interested in test the client. Please contact me
out of list [EMAIL PROTECTED] 

Thanks in advance.

--- Vlasis Hatzistavrou - asterisk mailing list
account [EMAIL PROTECTED] wrote:

 
  If anyone is interested I'm (slowly) developing a
 GPL'd Java applet that 
  works as an IAX softphone.
 
  I should have a test version out at the end of the
 week for a 
  limited number of testers.
 
  Tim.
 
 
  http://www.westhawk.co.uk/
 
 Hello Tim,
 
 We'd be interested to test the client...
 
 Best regards,
 Vlasis Hatzistavrou
 Technical Director  CEO
 Kinetix Tele.com Hellas Ltd.
 Monastiriou 9  Enotikon
 546 27
 Thessaloniki
 Greece
 Tel.: +302310556134
 Fax: +302310556134 (ext. 0)
 GSM: +306977835653
 e-mail: [EMAIL PROTECTED]
 http://www.kinetix.gr
 
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Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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Re: [Asterisk-Users] Minimum CPU required for 60 calls

On Tuesday 02 August 2005 09:55, Adam Dobrin wrote:
 And as much as you dislike these kinds of questions; its unfortunate
 that the community doesn't have any good answers to them available--they
 should be.  It would be great if we could get some independent
 verification of digium's claims/figures.

Unfortunately it's NOT that simple.  The motherboard and chipset play a big 
part in this, as does the network card, the types of codecs you're 
transcoding between, the SIP UAs if you're using it, what esle the box is 
doing...  

Honestly if you're unsure you grab a box you think will work (use some margin 
here) and try.  Report back.  It's the only way you'll figure it out.  I (and 
most others) can't be arsed to try and cut every corner so you just don't see 
the figures, and Digium's figures are for a specific system and setup.  YMMV.

-A.
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Re: [Asterisk-Users] Re: two UA with the same usr/pwd

2005-08-02 Thread Rich Adamson

Regardless of what has (or has not) been implemented in asterisk, there
is a very valid business reason for wanting an extension number to ring
on multiple phones and to determine the status of an extension from
multiple phones. Business have needed (and implemented) that for years.
Having such an implementation in asterisk would definitely be a major
plus (regardless of what our definitions of a pbx and keysystem happen
to be).


 Many people seem to want this feature.  I think they are just 
 confused.  I've never actually heard of a good reason to let multiple 
 devices register with the same username/secret.  Most of the time they 
   want a call to ring on multiple devices and they are trying to make 
 a device == extension, which is not correct.  A device is a device and 
 an extension is an extension and they are not the same thing and there 
 is no 1-to-1 mapping between them.
 
 Victor Alvarez wrote:
   I really think this matter deserves attention. I have been asked many 
  times about it.
  
   Regards,
Victor. 
  
  
 Hello,
 
  I can understand why asterisk is designed to not to allow two UAs with the 
  same usr/pwd, 
http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but 
I have to find a 
solution for this.
 
  My first option is use SER as an extension end of Asterisk, to allow more 
  than one SIP 
endpoint to register with the same details 
http://www.voip-info.org/wiki-Asterisk+at+large. I 
wonder if there is another way to do this. Of course, I am talking about a SIP 
proxy behaviour, 
simultaneous registration, both phones ringing at the same time and first to 
answer gets the 
call.
 
 Kind regards,
  Victor.
 
 
 
 -- 
 Always do right. This will gratify some people and astonish the rest.
 Mark Twain
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---End of Original Message-


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[Asterisk-Users] can one specify talking only for a participant in app_conference

2005-08-02 Thread Steven Langley









Hi there



I am wondering if anyone out there has used app_conference?
I am currently using meetme. My main problem with meetme is that one cannot
specify to stop sending voice packets to a participant when they are speaking.
This results in doubling the bandwidth for the participant compared to when
they are only listening.



In app_conference is it possible to stop sending voice
packets to a participant when they are talking?



Many thanks



Steven Langley






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[Asterisk-Users] Ztdummy or Zaptel card on production server

2005-08-02 Thread Steven Langley









Hi there



I am currently using Asterisk with Meetme on a 2.4 linux
kernel. I am using Ztdummy with usb-uhci driver for timing. It seems to work
ok, although I havent tried it with more than 5 users. However, I am now
looking to move into a production environment and some people have said that it
is better to have a Zaptel card. I would be installing Asterisk on a 2.6 linux
kernel for the production server, which wouldnt require usb-uhci if we
were to go for Ztdummy  but one person seemed to have found problems
with this: http://www.sineapps.com/print.php?rssid=731.
Also, if I were to go for a Zaptel card for timing, would pretty much any FXO
card be suitable?



Any advice on what to do would be great.



Steven Langley










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[Asterisk-Users] WHat does it take

2005-08-02 Thread Tim King









How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?








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Re: [Asterisk-Users] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx

2005-08-02 Thread Peter Svensson

On Tue, 2 Aug 2005, Frank Sautter wrote:

 Maik Schmitt schrieb:
 one of our customers which wants a soft transfer between his old pbx to
 asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
 everything is fine exept that when dialling from the legacy pbx it takes
 about 3 seconds before the asterisk start to dial.
 where does this delay come from?
 has it to do with 'overlapdial=yes'?
  
  This is normal behaviour if you use '.' in your extensions.conf. Use '!' 
  instead and Asterisk will start dialing immediately.

 when i change '.' to '!' then the overlap digits get lost. this means 
 the longest number dialled on my telco line is as long as there are 
 abigous matches in the dialplan.
 isn't there a way to start dialling after one received enough digits to 
 decide which path to dial and then still transmit the remaining 
 (overlapping) digits?

If you have overlap=yes on both legs this should work. What you are 
seeing is consistent with having overlap=no on the outgoing call leg.

Peter

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[Asterisk-Users] Can Asterisk Shoretel systems talk to each other?

2005-08-02 Thread Jimmy

(Please forgive my sending this again, but the list seems to have been 
acting up a little the last few days,  and I didn't see it appear when 
the list started working again)


We have a Shortel system at out main site.  We're putting Asterisk
servers at several smaller remote sites.  I know I'll be able to get the
Asterisk servers to talk to each other via IAX, but can they talk to the
Shoretel server?   Basically, I'd like to be able to, from the main site
with Shoretel, dial an extension, and reach that phone at a remote site,
and vice-versa.

Thank you for any input!

 -Jimmy Madden

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Re: [Asterisk-Users] Re: two UA with the same usr/pwd

2005-08-02 Thread Silik0nJesus (SwK)

On 8/2/05, Victor Alvarez [EMAIL PROTECTED] wrote:
 
  Hello,
  
   I can understand why asterisk is designed to not to allow two UAs with
 the same usr/pwd,
 http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html,
 but I have to find a solution for this.
  
   My first option is use SER as an extension end of Asterisk, to allow more
 than one SIP endpoint to register with the same details
 http://www.voip-info.org/wiki-Asterisk+at+large. I wonder
 if there is another way to do this. Of course, I am talking about a SIP
 proxy behaviour, simultaneous registration, both phones ringing at the same
 time and first to answer gets the call.
  
  I really think this matter deserves attention. I have been asked many times
 about it. 

If you really need 2 phones to ring at the same time and first to
answer get the call, all you have to do is Dial(SIP/phone1SIP/phone2)
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[Asterisk-Users] Has Sixtel gone under?

2005-08-02 Thread Carlos Chavez

 I have been using Sixtel from the beginning of the year and service was
getting worse and worse.  Yesterday I tried to access the website to get the
CDR and I got an error saying that the domain no longer exists.  I checked the
whois and it says that the domain is on hold.  Have they finally folded?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Sip over VPN not working

2005-08-02 Thread Tim P

Using the Xten X-Lite client (free) I am able to connect to a local
[EMAIL PROTECTED] server and when trying to connect to the remote server
(a mirror of the local) I am unable to connect.

The first server is a local lan, the remote is using microsofts pptp
vpn client to connect.

Looking at the diagnostics from X-Lite I see this:

begin log

SEND TIME: 341549001
SEND  192.168.8.151:5060
REGISTER sip:192.168.8.151 SIP/2.0
Via: SIP/2.0/UDP
192.168.8.203:5060;rport;branch=z9hG4bK3FB2D9058F8B41F79E68D710E384C12A
From: timtest sip:[EMAIL PROTECTED];tag=109208562
To: timtest sip:[EMAIL PROTECTED]
Contact: timtest sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 22204 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 341552005
SEND  192.168.8.151:5060
REGISTER sip:192.168.8.151 SIP/2.0
Via: SIP/2.0/UDP
192.168.8.203:5060;rport;branch=z9hG4bK3FB2D9058F8B41F79E68D710E384C12A
From: timtest sip:[EMAIL PROTECTED];tag=109208562
To: timtest sip:[EMAIL PROTECTED]
Contact: timtest sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 22204 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 341558014
SEND  192.168.8.151:5060
REGISTER sip:192.168.8.151 SIP/2.0
Via: SIP/2.0/UDP
192.168.8.203:5060;rport;branch=z9hG4bK3FB2D9058F8B41F79E68D710E384C12A
From: timtest sip:[EMAIL PROTECTED];tag=109208562
To: timtest sip:[EMAIL PROTECTED]
Contact: timtest sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 22204 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0

end log

My virtual IP (my vpn client) is 192.168.8.203 so that looks good and
the remote pbx is 192.168.8.151. I have checked and rechecked the
settings (extension, ips, password, display name) and certain they are
correct.

I don't see any activity on the asterisk console while the phone
attempts to register (I get the Login Failed, Contact Network Admin).
I have tried setting the Nat Firewall IP to both my virtual ip address
as well as the internal ip of the vpn server, no luck.

I am sure someone must have this setup in production if it is indeed
possible. Any suggestions?

thanks,
Tim
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[Asterisk-Users] Polycom phones w/ two lines on different servers

2005-08-02 Thread asterisk

Hi all -

This isn't really directly Asterisk related, but has anyone successfully
set up a Polycom phone to register two lines on two different Asterisk
boxes? I can get the first line to register, but the second one does not.
I can still place calls from that second line, which indicates to me the
server, user, and secret are correct. I'm running the newest 2.6 series
firmware with the newest SIP image.

My mac-sip.cfg contains:
phone1
   reg reg.1.displayName=1006 reg.1.address=[EMAIL PROTECTED]
reg.1.label=1006
   reg.1.type=private reg.1.auth.userId=1006 reg.1.auth.password=
   reg.1.server.dnsLookupOption=0 reg.1.server.1.address=192.168.0.90
   reg.1.server.1.port=5060 reg.1.server.1.expires=
   reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount=
   reg.1.server.1.expires.lineSeize= reg.2.displayName=1003
reg.2.address=[EMAIL PROTECTED]
   reg.2.label=1003 reg.2.type=private reg.2.auth.userId=1003
reg.2.auth.password=
   reg.2.server.dnsLookupOption=0 reg.2.server.1.address=192.168.2.2
reg.2.server.1.port=


The local overrides file for the phone contain:
voIpProt.server.2.expires.lineSeize=
voIpProt.server.2.retryMaxCount=
voIpProt.server.2.retryTimeOut=
voIpProt.server.2.register=
voIpProt.server.2.expires=3600
voIpProt.server.2.transport=DNSnaptr
voIpProt.server.2.port=5060
voIpProt.server.2.address=192.168.2.2

Any ideas? Sample configs would be appreciated if you've got something
like this working in the past.

Thanks,
Nick



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Re: [Asterisk-Users] Re: two UA with the same usr/pwd

2005-08-02 Thread Adam M. Dobrin

Rich Adamson wrote:

Regardless of what has (or has not) been implemented in asterisk, there
is a very valid business reason for wanting an extension number to ring
on multiple phones and to determine the status of an extension from
multiple phones. Business have needed (and implemented) that for years.
Having such an implementation in asterisk would definitely be a major
plus (regardless of what our definitions of a pbx and keysystem happen
to be).

I think you missed the point.

It is possible simple to do exactly what you want without using the
same username and password. Simply have the dialplan ring two different
phones..

Mine for instance, dials my work line, a sip phone at my home, and my
cell phone via zap simultaneously--you get much more control by having
separate devices.

-a

Many people seem to want this feature. I think they are just
confused. I've never actually heard of a good reason to let multiple
devices register with the same username/secret. Most of the time they
want a call to ring on multiple devices and they are trying to make
a device == extension, which is not correct. A device is a device and
an extension is an extension and they are not the same thing and there
is no 1-to-1 mapping between them.

Victor Alvarez wrote:

I really think this matter deserves attention. I have been asked many times
about it.

Regards,
Victor.

Hello,

I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd,

http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a
solution for this.

My first option is use SER as an extension end of Asterisk, to allow more than one SIP

endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I
wonder if there is another way to do this. Of course, I am talking about a SIP proxy behaviour,
simultaneous registration, both phones ringing at the same time and first to answer gets the
call.

Kind regards,
Victor.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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---End of Original Message-

[Asterisk-Users] Asterisk to Televantage

2005-08-02 Thread Sascha Ferley









Hi, 



We would like to hook a asterisk bases system into a
Artisoft Televantage based system. 
Does anyone have a idea of how to do this? Basically we need to trunk H323
lines. Is there anywhere a howto, or any documentation relating to this?



Please let me know

Thanks

Sascha








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[Asterisk-Users] Making a call on Asterisk... new thread or not?

2005-08-02 Thread Tim Karl


Hello,

Does anyone know how Asterisk manages calls on a system? More 
specifically, does it spawn a thread off of the asterisk program... are 
they separate processes? We're trying to see what kind of system load 
the PBX will create when calls are put through.


Thanks,

Tim
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Re: [Asterisk-Users] How to create a secret code to use [EMAIL PROTECTED] server's long distance plan from a public phone

2005-08-02 Thread asterisk


At 09:58 AM 8/2/2005, you wrote:

Adrien Laurent wrote:

Hello everyone,


I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.

When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.


Your going to have to add an extenion in the extensions_custom.conf for a 
DISA setup. Depending on what your extensions look like it's fairly easy 
to do.  Your system should be answered by the Digital Reception for this 
to work correctly or a dedicated DID routed to it. This is a quick sample.


lets give an extension like 300.

exten = 300,1,Goto(custom-disa,s,1)

[custom-disa]

exten = s,1,Authenticate(1234) ; any number you want for a password here.
exten = s,2,DISA,no-password|from-internal
exten = s,3,Hangup


How can I setup a secret password in the extension.conf, so that my
asterisk server can allow me to make long distance call ?


Thanks a lot,

Adrien


Since the OP is using AAH, I _think_ he can just change the context in the 
extensions setup screen to custom-disa. 


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Re: [Asterisk-Users] Re: two UA with the same usr/pwd

I ring multiple phones ALL THE TIME without needing duplicate 
username/secrets.  The following line wrapped, but you can still see 
what's happening.  When someone dials extension 3400 the devices with 
SIP the three SIP usernames (we set them to MAC-[a|b|c|d] where the 
letter indicates which line on the phone.)  We set each line on each 
phone to a seperate name.


exten = 
3400,1,Dial(SIP/0004f201862c-bSIP/0004f201fb0e-bSIP/0004f201e6a3-b,30)




Regardless of what has (or has not) been implemented in asterisk, there
is a very valid business reason for wanting an extension number to ring
on multiple phones and to determine the status of an extension from
multiple phones. Business have needed (and implemented) that for years.
Having such an implementation in asterisk would definitely be a major
plus (regardless of what our definitions of a pbx and keysystem happen
to be).



Many people seem to want this feature.  I think they are just 
confused.  I've never actually heard of a good reason to let multiple 
devices register with the same username/secret.  Most of the time they 
 want a call to ring on multiple devices and they are trying to make 
a device == extension, which is not correct.  A device is a device and 
an extension is an extension and they are not the same thing and there 
is no 1-to-1 mapping between them.



--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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RE: [Asterisk-Users] call center 20 seats

2005-08-02 Thread mattf

Hello,

You have several choices if you are doing almost all inbound, here's a
summary:
- Native Asterisk Agents and Queues (easy to setup but no screen pops
native. need add-ons for that, some are commercial)
- There are several companies that sell add-ons for Asterisk
queues/agents to extend functionality
- Aheeva Contact Center (all-in-one solution for in/out dialing with many
features, but it costs for install and maintenance)
- VICIDIAL (work with in/out and has screen pops, GPL and free, has many
features but not as well rounded as Aheeva)

As for trunks, I would recommend using IAX2 with something like a GSM or
G729 codec. There are many providers of IAX2 termination all over the world,
you should pick one that best fits your calling patterns and has the service
level you are looking for.
http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers

MATT---

-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 02, 2005 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center 20 seats


mattf wrote:

What kind of call center: inbound, outbound or both? 
  

It will be inbound 90%+ as I only need 2 seats for outgoing.

how many lines per agent will you have? 
  

one line per agent.

what kind of trunks will you be using?
  

Don't know yet. I am open for options and basically I don't want it to 
be bandwidth or process hungry.

do you need to tie into an existing database? 
  

No.

do you want screen-pops?

  

YES.

Let me know if you have any more questions.

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Re: [Asterisk-Users] WHat does it take

2005-08-02 Thread Doug Logan

It might be helpful if you posted your setup, and relative sections of your 
extensions.conf etc. 

Is this a new install? are you using VoIP extensions, FX, or what? Is the busy 
signal when you call from one extension to the other, when you dial-out? or all 
of the above?




Subject: [Asterisk-Users] WHat does it take
   From: Tim King [EMAIL PROTECTED]
   Date: Tue, 2 Aug 2005 11:28:37 -0400
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   asterisk-users@lists.digium.com

This is a multi-part message in MIME format.

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How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?

 


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Re: [Asterisk-Users] Full T38 sip Faxing now Available

2005-08-02 Thread Juan Jose Comellas

Please send this information to me also.


On Thu July 28 2005 01:03, Michael D Schelin wrote:
 Hello everybody, for all of you that have searched for a real fax
 solution, look no further. We now have T38 faxing. Please contact me for
 more information.

 Thanks

 Michael D. Schelin
 ShellTel
 626-814-2354



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-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] WHat does it take


I found your original message:

Can somebody please help here. At least respond and call me a moron. 
I have tried everything. I finally gave up and installed [EMAIL PROTECTED] 
from the iso and I am back to the exact same problem. Everything seems 
to work but my extensions are all busy. I used the AMP setup tool to 
add my zap extensions. If I view the console this is what happens when 
I call form one extension to the next. In the extensions setup when it 
asks what channel do I have to use a 2 digit number or something? This 
is a Digium TDM22B card. All the zaptel stuff seems to be working. And 
I can call out as well.


I must admit, I've not seen many messages quite as useless as this 
one.  First you are using [EMAIL PROTECTED], which few of us use.  Second 
you didn't bother to paste the output of the CLI for a failed call. 
You also did not post the relevant parts of your sip.conf or 
extensions.conf.  Unfortunatly posting these things are not going to 
be very useful since you are using [EMAIL PROTECTED] and I can't read 
their config files.



Tim King wrote:


How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?





 







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--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] WHat does it take


Tim King wrote:


How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?


Once is enough.  Perhaps you did not provide enough information for 
anyone to help you.  Perhaps your problem is compex enough that nobody 
can help.


There are many consultants around that will do Asterisk Work-For-Hire. 
 Digium also has paid support options.


Generally, I don't respond to people that post in HTML, people that 
don't write a readable and logical message, and people that ask the 
same question on the mailing lists over and over.  Actually that last 
catagory can get the person in my .procmailrc and have their messages 
automatically deleted.



--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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RE: [Asterisk-Users] WHat does it take

2005-08-02 Thread Geoff Manning




-Original Message-From: Tim King 
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 02, 2005 11:29 
AMTo: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: [Asterisk-Users] WHat does it 
take

How many times do you ask for help here before getting a 
respone? Every single thing I do No matter what I get busy extensions. I am 
willing to pay someone to help here. Anybody got a clue?


Why don't you resend the email request. We 
will need more info to help you out. The list just came back up yesterday after 
an outage of nearly a week so I'm sure everyone is just getting caught back up 
with the emails.
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Re: [Asterisk-Users] WHat does it take

2005-08-02 Thread Jon Pounder


 How many times do you ask for help here before getting a respone? Every
 single thing I do No matter what I get busy extensions. I am willing to
 pay
 someone to help here. Anybody got a clue?

go to the asterisk console, and do show dialplan, make sure things there
look as you expect from your config file.

then while you are in there initiate a call and see what extension it
tries to match to and verify there is one present in the right context
that matches. especially check the context of the call and where you
expect it to match are actually the same.

you should notice something from one of these places that doesn't seem as
you expect it so change that and repeat.

keep in mind on the mailing lists there is so much volume, people don't
have time to read every single email let alone take the time to reply.

I don't mind taking the time to answer someone quickly but I know I can't
do it for everyone, and I am sure most other people are in the same boat.
For those able to pay for help, there are lots of companies that can help
you.




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Jon Pounder

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Tools to Power Your e-Business Solutions
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[Asterisk-Users] TAPI driver: AstTAPI

2005-08-02 Thread Rodrigo Royo, Diego



Hello,

We are using AstTAPI to make call´s from Outlook and it woks perfect. Our issue 
is that it doesn´t work so well with our CRM. Although the TAPI line 
initialization is successful, AstTAPI doesn´t report the Phone Number and TAPI 
doesn´t know about our extension.  Anybody knows how to solve this?

Ours log:

2005/08/02 15:42:32:476 Information 0x50002 in Process 2040 Thread 1784
Location: CPHRequestManager::LineOpenWrapper Module: phreqmgr.cpp Line: 2385

The SAPphone server got a valid phone line. Parameters are: Line ID: 2 Line 
Name: Asterisk Line Handle: 66304 Number of Addresses: 1 Priviledges: 
0x0002 Media modes: 0x0004
-
2005/08/02 15:42:32:491 Information 0x50006 in Process 2040 Thread 1784
Location: CPHRequestManager::InitTapi Module: phreqmgr.cpp Line: 655

The TAPI line initialization was successful.  Line index: 2 Line handle: 66304 
Line version: 0x00020002

-
2005/08/02 15:42:36:162 Information 0x50007 in Process 2040 Thread 1784
Location: CPHRequestManager::InitTapi Module: phreqmgr.cpp Line: 747

The TAPI address initialization was successful.  Line index: 2 Address index: 0 
Phone Number: Unknown
-



Thanks,

Diego

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Re: [Asterisk-Users] priority a in macro to access voicemail

2005-08-02 Thread Moises Silva

i think may be you should read this:

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Macro

On 8/2/05, Damon Estep [EMAIL PROTECTED] wrote:
  
  
 
 I have added the following to a macro that is used for all extensions so a
 user can access voicemailmain by pressing * during the voicemail prompt 
 
   
 
 ; check voicemail 
 
 exten = a,1,voicemailmain(${macro_exten}) 
 
 exten = a,2,hangup 
 
   
 
 The behavior is a little weird, the * key is not recognized during the
 portion of the greeting where the extension number is being played back,
 after it is played back, for the duration of the greeting, the * key is
 recognized and works as expected. 
 
   
 
 Any ideas? 
 
   
 
   
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Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

2005-08-02 Thread Fredy Gonzales

I'm interested in test the client. Please contact me
out of list [EMAIL PROTECTED]

Gracias

FG

2005/8/2, Gonzalez Mata David [EMAIL PROTECTED]:
 I'm interested in test the client. Please contact me
 out of list [EMAIL PROTECTED]
 
 Thanks in advance.
 
 --- Vlasis Hatzistavrou - asterisk mailing list
 account [EMAIL PROTECTED] wrote:
 
 
   If anyone is interested I'm (slowly) developing a
  GPL'd Java applet that
   works as an IAX softphone.
  
   I should have a test version out at the end of the
  week for a
   limited number of testers.
  
   Tim.
  
  
   http://www.westhawk.co.uk/
  
  Hello Tim,
 
  We'd be interested to test the client...
 
  Best regards,
  Vlasis Hatzistavrou
  Technical Director  CEO
  Kinetix Tele.com Hellas Ltd.
  Monastiriou 9  Enotikon
  546 27
  Thessaloniki
  Greece
  Tel.: +302310556134
  Fax: +302310556134 (ext. 0)
  GSM: +306977835653
  e-mail: [EMAIL PROTECTED]
  http://www.kinetix.gr
 
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Re: [Asterisk-Users] TE410P

2005-08-02 Thread Michael B. Murdock

As a follow up to this thread..

We have been getting a number of cards now that have a PCI device id of
d161:0410. These all appear to be rev 2 digium cards and will not work
with version 1.0.7 (and previous) drivers. They require 1.0.9 drivers which
have this (and a few other) PCI device ids added to the table in wct4xxp.c.

Additionally, we are having PCI Parity errors (E13F4) on our Dell 2650's
with the rev 2 cards. Digium states that this is not a problem and can be
ignored but I am not sure my customers will agree. Has anyone else bumped
into this problem yet with the new Rev 2 cards?

-- Mike Murdock




- Original Message - 
From: Tony Hoyle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 07, 2005 5:21 PM
Subject: Re: [Asterisk-Users] TE410P


 Juan Pablo Abuyeres wrote:

  and there's nothing for vendor 79de at pcidatabase.com, and the kernel
  module loads well. Maybe there's more than one identifier for T410P ?
 
 There's nothing for 79de at pci-sig either
 (http://www.pcisig.com/membership/vid_search/search_form/process) and
 every PCI ID should be registered there...

 Probably a chinese manufacturer who just invented a number instead of
 properly allocating one.

 Tony
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[Asterisk-Users] Re: Minimum CPU required for 60 calls

2005-08-02 Thread M O

Adam,


I thought Andrew Kohlsmith gave the individual good
advice without intentionally malaciously spitting in
the guys face.

For the question,  'Whats the ' Minimum CPU required
for 60 calls? 

I think a Pentium 3, high end, which is cheap right
now, should do fine, but you will need either 3 T-1s
or 
arrange for the calls to come in via SIP, but you will
still need more than a DSL connection as you bandwidth
connection.

See also my replies below:

Message: 20
Date: Tue, 02 Aug 2005 09:55:20 -0400
From: Adam Dobrin [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Minimum CPU required
for 60 calls
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1;
format=flowed

And as much as you dislike these kinds of questions;
its unfortunate that the community doesn't have any
good answers to them available--they should be.  

Adam, join in and share your experiences.  

This is how this stuff grows :)  

It would be great if we could get some independent
verification of digium's claims/figures.

You brought up a valuable point, and I think, along
with you grabbing some hardware, you can help Verify
whatever Asterisk related questions you may have with 
the so called Digium.  LOL, they created it!

I mean, Adam, think that those Digium peoples threw
Asterisk out in the street without charging us a
penny.!  I'm with you, (NOT!), Digium owes us all

Relax dude, and check out here:

Dimensioning an Asterisk system

Typical questions asked on the mailing asterisk-users
are:

How fast/big must my machine be in order to serve my
needs?

How many simultaneous calls can Asterisk handle?

http://voip-info.org/tiki-index.php?page=Asterisk+dimensioning

#  (May 05) In my testing of CVS-HEAD I can get 5551+
sip sessions without media on asterisk without a
problem (testing with only 1 SIP user). The load
average is around 2-3. On a side note... on my 3ghz P4
HT box I can get 629 ulaw sip calls with media
(verified) without a problem. The load average on the
box was around 14 and it still sounded perfect... so
if you had a dual 3.4 ghz Xeon box you should have
ZERO problems doing a DS3 with asterisk. (That is if
the interrupt is 1000 per second and not 28000 and its
all ulaw) ... note that if you do not set the ulimit
-n 10 or something similar efore you start
asterisk you'll run out of FD's around 151 calls.

# (May 05) But the real scalability wall I've seen is
number of registered peers... That's what takes down a
box (at least with IAX). I've heard reports of a Dual
Xeon 3.2GHz not being able to handle even 1000 IAX
peers. ... I managed to get about 2500 users online on
one box by modifying iax2.h (reg expire changed from
from 60 to 240 seconds) decent IAX clients will comply
with that setting .. b.t.w. don't go higher, because
many NAT gateways will close their dynamic NAT
mappings after 300 secs, a few some even after 30
secs!

# (May 05) While you may not choose to put 10k users
on Asterisk, I have. Many more, as a matter of fact.
Some of these systems were simply media/application
servers, while some handled registrations as well.
While I agree that Asterisk needs some help on
registration volumes and scaling, I'd not sell it
short so quickly. At the moment, the only reason I
still would use SER would be for the registration and
call processing/loadbalance speed - Asterisk provides
all that I need for back-end call processing.

Adam, if the above is too much for you, you could
ALWAYS pay Signate $18,000 for the below description:

#  (Apr 05) Signate Telephony Server 5000#65533;s 51
Gigabits per second I/O capacity sustains more than
5,000 Session Initiation Protocol (SIP) call streams
per module using 80% of the capacity of a gigabit
network. Up to eight uniquely scalable modules can
share a 6.4GB/second interconnect based on SGI
technology that enables a coordinated system
supporting over 40,000 simultaneous call streams. 

Adam, for $18,000, this sounds like your Asterisk
machine.  I think I can build like 5 of these machines
for $18,000 ;)

The question itself is  Frustrating  as those in the
 Know  may not want to tell the world, but I have
heard that Platinum Tel, a prepaid seller of wireless
phones here in Chicago, uses Asterisk in some form of
GSM/CDMA gateway. :)  And no I dont know if it is true
or not, but you can always call them and ask.

You will only know the answer by  Hopping in the
swimming pool .  Remember, check out the
Voip-info.org wiki where it lists the Motherboards /
Hardware that creates issues for Asterisk.  

Then avoid those, and run with what is posted that
works like the above links.

For me, a Dell PowerEdge 2850 with 2 Gigs DDR2 Ram, 
and Dual Xeon 3.0Ghz processor with SCSI HD, RAID
Controller, and 2 GIGe Ethernet ports, should place me
at the 5500+ Sip concurrent sip calls.
 
Starting bandwidth for me, 100Mbps. ;)

I have Initial quantity of 20,000

Re: [Asterisk-Users] ast_config not updating voicemail password

2005-08-02 Thread Matthew Boehm


Bruce Komito wrote:

I've been using realtime to store my voicemail configuration in a mysql
table for several months now, and have had no problems...until today.  A
few weeks ago, I upgraded to the latest CVS and today I noticed voicemail
is not updating the password when the user changes it through option 0.
I'm not sure when this started happening, but I assume it was sometime
after I upgraded.

Has anyone else seen a problem like this, and if so, what's the solution?

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


There was a recent patch to voicemail that removed an incorrect error. 
It delt with changing password with realtime.


Does it say on console that voicemail password was changed? Does Allison 
say it was saved?


-Matthew

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[Asterisk-Users] Best way to connect asterisk to an traditional PBX

2005-08-02 Thread Administrator TOOTAI


Hi list,

we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). 
People from telco told that they can't connect two PBX's using E1/T1 or 
only with QSig signaling.


I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 
would be used. The PBX has a spare ADQ card installed on which we would 
connect. Has someone a such working setup? Is it working well?


More generaly, could you please tell me how you're connecting * to an 
traditional PBX, what you think is the best solution, which signaling 
you're using and which card(s).


At the moment, the two PBX's are connected through 2 TDM cards, one 4 
FXS the other 4 FXO. Our goal is to have a max of 30 lines available and 
available in the same time. Location is in France.


Thanks for your feedback.

--
Daniel
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[Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFO problems

2005-08-02 Thread Robert Murray




I have been playing 
with a 480i with the new firmware 1.2.0.162I hope to get some form 
of
paging intercom 
function to work. In the wiki someone post that ALERT_INFO type of paging 
might
be in this version 
of firmware but I have been unable to find anything on this 
yet.

I have tried sending 
the ALERT_INFO to the phone a number of ways with no results. I then hooked 
up
my bt100 and tried 
to dial my "page" extension and then I started to see 403 error reporting on the 
bt100
using sip debug it 
looks like the 480i is returning the 403 message.

Am not sure if this 
is telling me that ALERT_INFO is not doing anything for now in the 480i or if I 
just don't know what
string to send to 
the phone in the ALERT_INFO message. 

I assume that 
ALERT_INFO is just not set up to do anything with the 480i for now. But I was hoping 
someone else
might know or have 
had luck with getting ALERT_INFO messages with the 480i working.


Robert
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Re: [Asterisk-Users] WHat does it take