Re: [Asterisk-Users] Best way to automatically stop and start Asterisk nightly

[Jean-Michel Hiver]

Chuck Bunn a écrit :


Hi,

I am planning on restarting asterisk nightly as I seem to be 
experiencing some sort of memory leak (Asterisk slows down over time). 
I have reviewed the Asterisk suggestions for management and one item 
is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what 
is the recommend way to implement an automatic stop and start of 
asterisk (there are changes in 1.2 with reload and restart) and is 
this enough or should I restart the hardware as well??


I'd use asterisk -rx 'restart when convenient'


Cheers,
Jean-Michel.

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[Asterisk-Users] Can Asterisk replace Cisco 5350?

[Linuxnizer The Mesmorizer]

Hi,
 We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1) 
to terminate calls. We need 2 more gateways, my question is can we save some 
money and use Asterisk + PCI E1 cards?


If so, do you recommend any cards/configuration?

Thank you
Ahmed

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Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?

[Jean-Michel Hiver]

Linuxnizer The Mesmorizer a écrit :


Hi,
 We are using Cisco5350 as a gateway with 2 E1 cards (part# 
AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my 
question is can we save some money and use Asterisk + PCI E1 cards?


I've had the same issue lately. I need to set up a 4E1 / g.729 solution.


Asterisk way


- 4 asterisk boxes with 1 E1 card (approx $2k each)
- 120 g.729 licences ($1.2k)
- 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k)

Total: 4 * 2 + 1.2 + 1 = 10.2


In the end, I went on voipsupply.com and saw that they offer Audiocodes 
mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 
(which is nice if you want to properly interconnect some day), can scale 
up to 16 E1 and is conveniently packed in a 1U rackable unit, I have 
decided to go with Audiocodes.


Since I am not set up yet, I can't tell wether it is a good decision or 
not. I will let you know :)



Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Re: Does hardware like this exist...?

[Erwin de Raad]

 BJ Weschke wrote:
  On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote:
 
 Hi all!
 
 I am looking for a device that I can stick in a USB-port on my Asterisk
server and that allows me to connect one/more (cordless) PSTN-phones in such
a way that they'll work with SIP/Asterisk. I know
 there are USB-phones, but what I'm looking for is 'the USB-phone without
the phone', if you know what I mean...   ;-)
 

Around Christmas last year I ordered a VTA1000 from
http://www.pcphoneline.com
It uses a Windows app so not sure (or tried...) how this works on linux.

Mine is the VTA1000-Skype and I tied it to a SPA3000. This is my Skype to *
gateway.

Erwin

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RE: [Asterisk-Users] CID lookup from an Exchange Public folder

[Steve Hanselman]

We have a public folder full of contacts, but I understood that you
could only access this if the contacts were contacts in AD?

I was planning on doing a match on telephone number, mobile number and
fax.  And then pulling a shortened version of the name as the caller ID,

Steve


-Original Message-
From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 16 December 2005 21:48
To: asterisk-users@lists.digium.com
Cc: Steve Hanselman
Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder

Steve,

You can get to anything in Exchange via LDAP. What is and or is not
working? Where are you entering the callerID info you want pulled?
Please see attachment for where you might want to enter this. Please
share if you get this.

Cheers.

Jason


-
Message: 1
Date: Fri, 16 Dec 2005 18:29:12 -
From: Steve Hanselman [EMAIL PROTECTED]
Subject: [Asterisk-Users] CID lookup from an Exchange Public folder
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Has anybody done this?

I looked at LDAP but you can't get to them that way, I'm considering
either a timed export, or some other way (can you access them via IMAP?
Or by wget on the owa web structure?)

Steve




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Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?

[AR Tarzi]

Ignoring SS7, why exactly are you setting up several boxes ? there are quad 
E1 cards no ?

This is way out of my league, but I just want to understand.

- Original Message - 
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Saturday, December 17, 2005 12:19
Subject: Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?



Linuxnizer The Mesmorizer a écrit :


Hi,
 We are using Cisco5350 as a gateway with 2 E1 cards (part# 
AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question 
is can we save some money and use Asterisk + PCI E1 cards?


I've had the same issue lately. I need to set up a 4E1 / g.729 solution.


Asterisk way


- 4 asterisk boxes with 1 E1 card (approx $2k each)
- 120 g.729 licences ($1.2k)
- 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k)

Total: 4 * 2 + 1.2 + 1 = 10.2


In the end, I went on voipsupply.com and saw that they offer Audiocodes 
mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 
(which is nice if you want to properly interconnect some day), can scale 
up to 16 E1 and is conveniently packed in a 1U rackable unit, I have 
decided to go with Audiocodes.


Since I am not set up yet, I can't tell wether it is a good decision or 
not. I will let you know :)



Cheers,
Jean-Michel.

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Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?

[Simone Cittadini]

Matt Florell ha scritto:


The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many others on this list) when trying to
use them in production for Asterisk. Take a look at the Digium
compatibility list:
http://www.digium.com/index.php?menu=compatibility
 

*I've installed a PowerEdge 2850 - Xeon 3.0GHz/2MB, 800FSB with two 
TE410P in it, the cards didn't worked out of the box, but they worked 
after a couple of hours googling around, and it is in production since 3 
months, never gone down.

*

*(I'm not advocating dell, actually I don't even like dell as a society, 
only sharing my experience)

*


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Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?

[Jean-Michel Hiver]

AR Tarzi a écrit :

Ignoring SS7, why exactly are you setting up several boxes ? there are 
quad E1 cards no ?

This is way out of my league, but I just want to understand.


Because you would need a super monster box to do simultaneous g.729 
encoding - and even though I'm not sure it would work properly. Maybe 
when we have boards which support hardware g.729 encoding this will 
become a viable option.


Cheers,
Jean-Michel.

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[Asterisk-Users] I need syntax on applicationmap in features.conf

[Obelix]

I need some information on the syntax used in features.conf.

I want to use the applicationmap to assign different buttons to the Hangup()
command. Where should I look?

Obelix

  I want to use '##' to terminate a call instead of the '*' used by the Dial
  command's H option.
 
  Is there a way to change the key or use another option to achieve the same
  effect?

 Application map in features.conf assigning ## to Hangup() ?

 Maybe :)

 --
 Cheers,



This message was sent using IMP, the Internet Messaging Program.

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Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?

[Linuxnizer The Mesmorizer]

From: Jean-Michel Hiver [EMAIL PROTECTED]

Linuxnizer The Mesmorizer a écrit :


Hi,
 We are using Cisco5350 as a gateway with 2 E1 cards (part# 
AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question 
is can we save some money and use Asterisk + PCI E1 cards?


I've had the same issue lately. I need to set up a 4E1 / g.729 solution.


Asterisk way


- 4 asterisk boxes with 1 E1 card (approx $2k each)
- 120 g.729 licences ($1.2k)
- 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k)

Total: 4 * 2 + 1.2 + 1 = 10.2


In the end, I went on voipsupply.com and saw that they offer Audiocodes 
mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 
(which is nice if you want to properly interconnect some day), can scale up 
to 16 E1 and is conveniently packed in a 1U rackable unit, I have decided 
to go with Audiocodes.


Since I am not set up yet, I can't tell wether it is a good decision or 
not. I will let you know :)



Cheers,
Jean-Michel.


Hii Jean-Michel,
 Couple of notes, I didn't find Audiocodes at voipsupply.com. As far as the 
E1 is concerned, I think that there are many standards for R2-E1 signaling. 
Cisco support many variations, not sure if these cards or Asterisk support 
such wide variaty of R2 signaling. Check Cisco paper on this 
http://www.cisco.com/en/US/products/sw/iosswrel/ps1826/products_feature_guide09186a00800dc5cf.html


Final note, I can get a used Cisco5350 for around $7000 with 2E1 cards, your 
solution seems a bit pricey compared to a used Cisco. Any advantages or 
features that come with Asterisk that can't be done with a Cisco5350?


Regards,
Linuxman.

_
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7.5 today! http://messenger.msn.co.uk


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[Asterisk-Users] Key R (Flash) and Asterisk

[Linux Administator]

Hi
I need send a codenumber + key R (flash) from isdn telephone to a interface
on pstn.


isdn telephone -asterisk - (fxo)-- interface

Help me!!!

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Re: [Asterisk-Users] Can Asterisk replace Cisco 5350?

[Jean-Michel Hiver]

Hii Jean-Michel,
 Couple of notes, I didn't find Audiocodes at voipsupply.com.


This is the product I'm going to order:

http://www.voipsupply.com/product_info.php?products_id=213osCsid=8afe5c480fd75d05ce6e5dad5876e3be


Final note, I can get a used Cisco5350 for around $7000 with 2E1 
cards, your solution seems a bit pricey compared to a used Cisco. Any 
advantages or features that come with Asterisk that can't be done with 
a Cisco5350?


I don't know Cisco enough to be able to compare.

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context --ResponseTimeout the best answer?

[Philipp von Klitzing]

Hi!

Upgarde to 1.2.1 and try again - 1.2.0 (and maybe the beta) had a bug 
concerning .call files and the non-passing on of variables that might 
affect you as well.

Cheers, Philipp

 Hmmm seems like every dialplan snippet I've seen so far relies on
 ResponseTimeout and looping back to s,1. Is this the only way I can get this
 to work kind-of the way I want? Any ideas welcome. 

 Weird thing is, I swear this worked the way I wanted it to when I was
 running 1.0.9, now it's not under 1.2 beta 1. Am I crazy?


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[Asterisk-Users] asterisk 1.2.1 realtime mysql.4.1.xx report errors

[hoowa sun]

i am using asterisk1.2.1 realtime mysql4.1.x

i found same update error in debug mode

i cat /var/log/asterisk/debug follow:Dec 13 00:12:28 DEBUG[7533] db.c: Unable to find key '99015' in family 'SIP/Registry'Dec 13 00:12:28 DEBUG[7533] res_config_mysql.c: MySQL RealTime: Update SQL: UPDATE sip_user SET ipaddr = '', port = 'Ð'ê·¬'ê·qœ¥', regseconds = '0', username = '99015' WHERE name = '99015'
Dec 13 00:12:28 DEBUG[7533] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_user SET ipaddr = '', port = 'Ð'ê·¬'ê·qœ¥', regseconds = '0', username = '99015' WHERE name = '99015'Dec 13 00:12:28 DEBUG[7533] res_config_mysql.c: MySQL RealTime: Query Failed because: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'ê·¬'ê·qœ¥', regseconds = '0', username = '99015' WHERE name = '99015'' at line 1
Dec 13 00:12:28 DEBUG[7529] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_user WHERE name = '99015'Dec 13 00:12:28 DEBUG[7529] db.c: Unable to find key '99015' in family 'SIP/Registry'

this UPDATE statements oftenget error messages.
arethose bug???
how areinfluence on my services?

thanks
hoowa sun


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Re: [Asterisk-Users] Re: Codecs.

[Rich Adamson]

   Hi all i have some problems with my pbx and asterisk codecs.
   
   if i use g711u or g711a codecs. the line never hangup. and the origin
   and destination are connected until i restart my pbx or asterisk
   
   But if i use GSM all work fine.
   
   is possible to solve this problem? or use only gsm codec?
  
 
  Yes, its possible to solve the problem.

 can you explain how?

Not without you providing at least something to give us a clue what it
is that you've programmed into your system. 

How about if you give us some clue as to which version of * you're
using, what type of phones are associated with origin and destination,
if these are sip phones what do your sip.conf definitions look like,
what does the appropriate sections of extensions.conf look like, and
any other configuration pieces that might pertain to whatever it is
that you've implemented. Your posting implies there might be more than
one * system involved and possibly even iax trunking, etc.


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RE: [Asterisk-Users] A2billing Trunk

[Steve Totaro]

 
 Excuse me Chris!
 
 Forgive me that I don't understand what you are really mean?
 
 I would very appreciated if you let me know some think about the
rules,
 and
 how we would get help from people and how to find some previous
 information
 that has been posted from someelse before that we may need it.
 
 I think we need to create a knowledge base database, so members can
come
 in
 to search and share the information that we need?
 
 Thanks!
 
 Lan.
 
 


There a couple REALLY good knowledge bases you can check for your
answers.  Here is one specific for your problem.
http://www.areski.net/a2billing/html/book1.html
Next I suggest:
http://www.voip-info.org
And the best I have found so far:
www.google.com

Thanks,
Steve
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Re: [Asterisk-Users] TDM01B answering issue

[Rich Adamson]

 Is there any possible way to make TDM01B  answers when
 the other side pick up the phone ,and to prevent it
 from answering just when it starts ringing?

Yes, if I understand your question properly.

Suggest you post relavent parts of zapata.conf and extensions.conf
that are associated with the fxo port of the TDM card. No one can
help you unless you provide _some_ detail how you have this stuff
configured.


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[Asterisk-Users] Cid_rewrite update

[Technical Support]

I suspect lots of people use the cid_rewrite script by Jay Milk.  It's a
great script that updates the CID info by looking up callerid ID from
411.com (reverse lookup)

The script seems to be stuck at version 1 so I added a few enhancements to
bring it up to ver 1.1  The biggest is the addition of a column to the
callerid MYSQL table called action.  The single character is:

R = Restricted (allow access during set hours)
N = Nuisance (block this caller)
U = Unrestricted (always allow access)

This is returned to the asterisk dialplan as a variable called ACTION.

Not a big deal, but a nice start!  Does anyone else of
updates/recommendations I should roll in before I release this script?

Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your busine
E.
 
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca 


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Re: [Asterisk-Users] FOP button limit?

[Nicolás Gudiño]

 50 extensions, 27 trunks, 1 queue, any tips would be great appreciated,
 -Kerry

Inside op_style.cfg:

btn_width=191
btn_height=30
btn_padding=5

Then tweak all the scales and margin parameters for the icons. It
would give you all the buttons you need an a couple more.

You can direct all this questions to FOP's mailing list, you can
subscribe from http://www.asternic.org or browse the archive for some
style examples. Regards,

--
Nicolás Gudiño
Buenos Aires - Argentina
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[Asterisk-Users] multiple ALSA devices and Asterisk

[Dan]

Hi all,

There is any possibility to have two local consoles using ALSA 
devices?

I see no such an option in the alsa.conf nor extensions.conf files

Thank you and best regards,
Dan


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[Asterisk-Users] Re: [Astguiclient-users] [PATCH][RFC] Quiet debugging messages in Net::MySQL Perl module

[Mike Fedyk]

That TINYINT is probably the culprit then since the message is in 
short.  That code is converting the number to a 16bit short value.  Are 
there any other perl scripts that modify tables with TINYINTs in them?


From looking at the module, it doesn't look like it is reporting an 
error, but just outputting when doing the number conversion (going to 
16bit can be lossy).


Do we really have an issue here besides quieting the debug message?

Mike

mattf wrote:


Hello,

No, alphamuneric is not strange, we have letters and numbers in ours and
I've never had an issue with it.

There is one TINYINT field for dial_timeout, other than that it's all
VARCHARs and ENUMs.

MATT--

-Original Message-
From: Mike Fedyk [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 17, 2005 3:45 AM
To: mattf
Cc: [EMAIL PROTECTED]
Subject: Re: [Astguiclient-users] [PATCH][RFC] Quiet debugging messages
in Net::MySQL Perl module


mattf wrote:
 


looks like this is the statement that causing problems:
SELECT * from vicidial_campaigns where active='Y' or
SELECT * from vicidial_campaigns where campaign_id='$CLIcampaign'

I have no idea how that would return an error, What are your campaign
   


names?
 


do you have any strange character in them?
 
   


Is [a-zA-Z0-9] strange?

I suspect it is converting data for numerical fields.  Are there any INT 
fields in that table?


I did a google for that error, but nobody has debugged it.

Mike


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[Asterisk-Users] Cisco 79xx display as busy-lamp field

[Bruce Komito]

Has anyone used a Cisco 7940/7960 (with or without a 7914) to display busy
extensions and if so, would you mind sharing the XML code to do it?

TIA

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


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[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Evil Skymarshal]

Hi,I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For testing reasons I but the following in extensions.conf---cut---[from-sip]exten = 2000,1,Answer()exten = 2000,2,Wait(1)
exten = 2000,3,SayDigits(123)exten = 2000,4,Hangup()---cut---When ever I call the 2000 asterisk -vc says:---cut--- -- Executing Answer(SIP/2303-1ae1, ) in new stack
 -- Executing Wait(SIP/2303-1ae1, 1) in new stack -- Executing SayDigits(SIP/2303-1ae1, 123) in new stack -- Playing 'digits/1' (language 'en') -- Playing 'digits/2' (language 'en')
 -- Playing 'digits/3' (language 'en') -- Executing Hangup(SIP/2303-1ae1, ) in new stack---cut---BUT I don't hear it everytime! Why? Sometimes I can hear it and most time I can not. I redial 20 times and I can hear it only 3-4 times. Does anybody have an idea what kind of strange problem that could be?
Thanx ES
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[Asterisk-Users] Alarm panel through ATA

[Chris Mason (Lists)]

I am trying to get some feedback from anyone who may have experience of 
a problem I am having.
We have several buildings that having only fiber to them so in order 
that the alarm panel can call the central station, I have provided a 
Sipura 1001 ATA. I can make a call to the central station through the 
ATA, using an analogue phone.
The alarm panel does not seem to function properly thorugh the ATA, 
either it is not going off-hook properly or the ATA is treating the 
modem tones as a fax, I am not sure what is happening. Does anyone have 
experience of getting this to work?


--
Chris Mason


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[Asterisk-Users] i can't register to my sip service(but x-lite can)

[hoowa sun]

i can't register to my sip service.but x-lite can.
i think because my sip service domain is not really domain, they using sip proxy to resolve this domain

who can help me fix this problem thanks :)

look for follow line:

Asterisk SIP REGISTER header
-

REGISTER 12 headers, 0 linesReliably Transmitting (no NAT) to 222.36.0.13:5060:REGISTER sip:222.36.0.13 SIP/2.0Via: SIP/2.0/UDP 
220.201.97.133:6699;branch=z9hG4bK330cece3;rportFrom: sip:[EMAIL PROTECTED];tag=as20630857To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]CSeq: 102 REGISTERUser-Agent: AsteriskMax-Forwards: 70Expires: 120Contact: 
sip:[EMAIL PROTECTED]:6699Event: registrationContent-Length: 0
- SIP read from 222.36.0.13:5060:SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 220.201.97.133:6699;branch=z9hG4bK330cece3;rport
From: sip:[EMAIL PROTECTED];tag=as20630857To: sip:[EMAIL PROTECTED];tag=2129750016Call-ID: 
[EMAIL PROTECTED]CSeq: 102 REGISTERContent-Length: 0

X-LITE REGISTER HEADER

SEND TIME: 25541500SEND  222.36.0.13:5060REGISTER sip:test.com SIP/2.0Via: SIP/2.0/UDP 221.201.158.56:5060
;rport;branch=z9hG4bKD0D7DC611385496CB06400989CF1AC0CFrom: 99970206 sip:[EMAIL PROTECTED];tag=1171724134To: 99970206 
sip:[EMAIL PROTECTED]Contact: 99970206 sip:[EMAIL PROTECTED]:5060Call-ID: [EMAIL PROTECTED]
CSeq: 1484 REGISTERExpires: 1800Max-Forwards: 70User-Agent: X-Lite release 1103mContent-Length: 0
RECEIVE TIME: 25541859RECEIVE  222.36.0.13:5060SIP/2.0 200 OKVia: SIP/2.0/UDP 221.201.158.56:5060;rport;branch=z9hG4bKD0D7DC611385496CB06400989CF1AC0C
From: 99970206 sip:[EMAIL PROTECTED];tag=1171724134To: 99970206 sip:[EMAIL PROTECTED];tag=1650035160Call-ID: 
[EMAIL PROTECTED]CSeq: 1484 REGISTERContact: 99970206sip:[EMAIL PROTECTED]:5060Expires: 30Content-Length: 0

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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

What are you codec and dmtfmode settings in sip.conf and in the sip 
phone settings. If you dmtfmode is set to 'inband' and you are using 
anything other than ulaw or alaw codec it wont work. Also since your 
hear the phone sometimes you may be experiencing QOS issues on your 
network. Doe you have QOS set up on your switches in the points between 
the server running asterisk and the sip client?


Hope this helps

Evil Skymarshal wrote:


Hi,

I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For 
testing reasons I but the following in extensions.conf


---cut---
[from-sip]
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,SayDigits(123)
exten = 2000,4,Hangup()
---cut---

When ever I call the 2000 asterisk -vc says:

---cut---
-- Executing Answer(SIP/2303-1ae1, ) in new stack
-- Executing Wait(SIP/2303-1ae1, 1) in new stack
-- Executing SayDigits(SIP/2303-1ae1, 123) in new stack
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Executing Hangup(SIP/2303-1ae1, ) in new stack
---cut---

BUT I don't hear it everytime! Why? Sometimes I can hear it and most 
time I can not. I redial 20 times and I can hear it only 3-4 times. 
Does anybody have an idea what kind of strange problem that could be?


Thanx
  ES



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Re: [Asterisk-Users] Alarm panel through ATA

[Gonzalo Gonzalez]

I just had a setup like that;  the alarm company is coming next week to
install and test.  Make sure the panel is setup for DTMF and not for pulse;
I have found this is the case on some panels.

Gonzalo Gonzalez


- Original Message - 
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: Asterisk-Users asterisk-users@lists.digium.com
Sent: Saturday, December 17, 2005 11:20 AM
Subject: [Asterisk-Users] Alarm panel through ATA


 I am trying to get some feedback from anyone who may have experience of
 a problem I am having.
 We have several buildings that having only fiber to them so in order
 that the alarm panel can call the central station, I have provided a
 Sipura 1001 ATA. I can make a call to the central station through the
 ATA, using an analogue phone.
 The alarm panel does not seem to function properly thorugh the ATA,
 either it is not going off-hook properly or the ATA is treating the
 modem tones as a fax, I am not sure what is happening. Does anyone have
 experience of getting this to work?

 -- 
 Chris Mason


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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Evil Skymarshal]

Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sipphone settings.I use gsm.
 If you dmtfmode is set to 'inband' and you are usinganything other than ulaw or alaw codec it wont work.I changed the settings and tried:---cut---exten = 2000,1,Answer()exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)exten = 2000,4,Hangup()---cut---Same problem. Sometimes it works but most of the times it doesn't.
 Also since yourhear the phone sometimes you may be experiencing QOS issues on yournetwork.Of course it could be a QOS problem. But should I hear at least something?cu ES

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Re: [Asterisk-Users] Alarm panel through ATA

[Rich Adamson]

 I am trying to get some feedback from anyone who may have experience of 
 a problem I am having.
 We have several buildings that having only fiber to them so in order 
 that the alarm panel can call the central station, I have provided a 
 Sipura 1001 ATA. I can make a call to the central station through the 
 ATA, using an analogue phone.
 The alarm panel does not seem to function properly thorugh the ATA, 
 either it is not going off-hook properly or the ATA is treating the 
 modem tones as a fax, I am not sure what is happening. Does anyone have 
 experience of getting this to work?

Is the alarm panel truly expecting to use a modem to communicate with
central station?

If so, sip/rtp will not handle modems that attempt to use anything
greater then about 2400 baud. (Note: there is a diffence is the term
baud verses bits/second. Newer high-speed modems use an encoding
mechansim that involves phase-shift technology to achive a higher
bit-per-second speed over a low baud rate. sip/rtp will not accurately
reproduce any modem signal that involves phase-shifting. The sampling
rate is not sufficient to accurately reproduce phase-shifted analog
signals.)

Also, in the sipura release notes (for v3.1.5) specifically
indicates they watch for fax tones and, more recently, modem tones.

.Distinguish between FAX Passthrough mode and Modem Passthrough Mode.
 Modem Passthrough Mode can only be triggered by predialing the
 Modem Line Toggle Code. FAX Passthrough Mode is triggered by
 CED/CNG tone or NSE events. Echo canceller is automatically disabled
 for Modem Passthrough Mode only. Echo canceller is automatically disabled
 only if FAX Disable ECAN (Line 1/2) is set to yes for that line
 (in that case FAX passthrough is the same as Modem passthrough).
 Call-waiting and silence suppression is automatically disabled for
 both FAX and Modem passthrough as before. In addition, out-of-band 
 DTMF Tx is disabled during modem or fax passthrough (all audio are

The Modem Line Toggle Code: option appears on the Regional tab and
has a default string of *99. Therefore, your alarm panel modem would
need to prefix its dialing with *99.

Might check those two items.

Rich


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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

If you do not have QOS assigned to the SIP protocol it is quite possible 
that there are packets time outs and the packets are discarded. Is it 
possible to test the network during the evening or at a time when 
traffic is at it lowest? Also try several traceroutes and see if there 
is a wide variation in return times (widely varying treceroutes could 
indicate network saturation). You are using gsm are you using 
dmtfmode=rfc2833 or something else (this must be set in the sip.conf and 
on the sip soft phone and they must match!)


Thanks

Evil Skymarshal wrote:


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.


I use gsm.

If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.
 


Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network.


Of course it could be a QOS problem. But should I hear at least something?

cu
  ES



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Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
 



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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

Something else I should mention. Sip uses UDP and TCP packets. TCP 
packets are used if there is congestion on the network. I am unclear 
about what mechanism causes sip to switch between UDP and TCP but I 
believe it is controllable - I believe It would be easier to use QOS 
though. If UDP is used that packets could time out and you would never 
know it since UDP is dumb and has no packet loss recovery mechanism. 
What is the topology of your network. Is the Asterisk box and the client 
on the same backbone and switch?


Thanks

Evil Skymarshal wrote:


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


What are you codec and dmtfmode settings in sip.conf and in the sip
phone settings.


I use gsm.

If you dmtfmode is set to 'inband' and you are using
anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.
 


Also since your
hear the phone sometimes you may be experiencing QOS issues on your
network.


Of course it could be a QOS problem. But should I hear at least something?

cu
  ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
 



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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Rich Adamson]

I don't believe asterisk has any sip tcp support. Its all udp.


 Hi,
 
 Something else I should mention. Sip uses UDP and TCP packets. TCP 
 packets are used if there is congestion on the network. I am unclear 
 about what mechanism causes sip to switch between UDP and TCP but I 
 believe it is controllable - I believe It would be easier to use QOS 
 though. If UDP is used that packets could time out and you would never 
 know it since UDP is dumb and has no packet loss recovery mechanism. 
 What is the topology of your network. Is the Asterisk box and the client 
 on the same backbone and switch?
 
 Thanks
 
 Evil Skymarshal wrote:
 
  Hi Chuck,
 
  2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED]:
 
  What are you codec and dmtfmode settings in sip.conf and in the sip
  phone settings.
 
 
  I use gsm.
 
  If you dmtfmode is set to 'inband' and you are using
  anything other than ulaw or alaw codec it wont work.
 
 
  I changed the settings and tried:
  ---cut---
  exten = 2000,1,Answer()
  exten = 2000,2,Wait(1)
  exten = 2000,3,Playback(hello-world)
  exten = 2000,4,Hangup()
  ---cut---
 
  Same problem. Sometimes it works but most of the times it doesn't.
   
 
  Also since your
  hear the phone sometimes you may be experiencing QOS issues on your
  network.
 
 
  Of course it could be a QOS problem. But should I hear at least something?
 
  cu
ES
 
 
 
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 No virus found in this incoming message.
 Checked by AVG Free Edition.
 Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005
   
 
 
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---End of Original Message-


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Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

Rich I stand corrected you are absolutely right - see 
http://www.voip-info.org/wiki-Asterisk+config+sip.conf


The following appears on the page:


   Please note

   * Asterisk does not yet support SIP over TCP. It only supports SIP
 http://www.voip-info.org/wiki/view/SIP over UDP.
   * For Grandstream http://www.voip-info.org/wiki/view/Grandstream
 phones: set *dtmfmode=info*
   * Asterisk uses the incoming RTP
 http://www.voip-info.org/wiki/view/RTP Stream as a timing source
 for sending its outgoing Stream. If the incoming stream is
 interrupted due to silence suppression then musiconhold will be
 choppy. So in conclusion, you cannot use silence suppression.
 *Make sure ALL SIP phones have disabled silence suppression.*
 There is a solution for the silence suppression problem, see bug
 5374 http://bugs.digium.com/view.php?id=5374 for details.

Thanks


Rich Adamson wrote:


I don't believe asterisk has any sip tcp support. Its all udp.


 


Hi,

Something else I should mention. Sip uses UDP and TCP packets. TCP 
packets are used if there is congestion on the network. I am unclear 
about what mechanism causes sip to switch between UDP and TCP but I 
believe it is controllable - I believe It would be easier to use QOS 
though. If UDP is used that packets could time out and you would never 
know it since UDP is dumb and has no packet loss recovery mechanism. 
What is the topology of your network. Is the Asterisk box and the client 
on the same backbone and switch?


Thanks

Evil Skymarshal wrote:

   


Hi Chuck,

2005/12/17, Chuck Bunn [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


   What are you codec and dmtfmode settings in sip.conf and in the sip
   phone settings.


I use gsm.

   If you dmtfmode is set to 'inband' and you are using
   anything other than ulaw or alaw codec it wont work.


I changed the settings and tried:
---cut---
exten = 2000,1,Answer()
exten = 2000,2,Wait(1)
exten = 2000,3,Playback(hello-world)
exten = 2000,4,Hangup()
---cut---

Same problem. Sometimes it works but most of the times it doesn't.


   Also since your
   hear the phone sometimes you may be experiencing QOS issues on your
   network.


Of course it could be a QOS problem. But should I hear at least something?

cu
 ES



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005


 


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---End of Original Message-






 



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[Asterisk-Users] Terminating calls externally via SER

[Douglas Garstang]

I'm wondering if anyone has ever implemented a scenario where calls aren't 
terminated directly via Asterisk, but instead are passed back to a proxy, such 
as SER to terminate the calls. With basic dialling, it would be easy. For basic 
calling...
 
exten = XXX, 1, Dial(SIP/[EMAIL PROTECTED],20,tr)
 
What about more advanced stuff tho? With Queues, the AgentCallbacklogin command 
requires as an arguement where to reach the extension AND you have to supply 
the context. Would Asterisk take something like:
 
exten = 8000,1,AgentCallBackLogin([EMAIL PROTECTED]@proxy)
 
Don't know if Asterisk would correctly parse two @ symbols. Also, what about 
hints? Would something like this be needed?
 
5551212 = hint,1,SIP/[EMAIL PROTECTED]
 
Would Asterisk even accept that? And so on. I guess I need to know if all the 
Asterisk commands that require an extension can have a @proxy prefixed to the 
end, and still work (assuming 'proxy' is defined in sip.conf of course).
 
Thanks.
Doug
 
 
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Re: [Asterisk-Users] Alarm panel through ATA

[Jerry Jones]

This is a very big headache for me.

Alarms today normally use a protocol called contactID for  
communications. These are very short dtmf tones and most devices have  
a very hard time transmitting them. also any jitter etc causes them  
to be unreadable. If anyone has a reliable method for transporting  
across an IP net I would like to hear about it.


The only solution I have found, and the manufacturers seem to agree  
is to not use contactid. Most central alram stations support a couple  
others. One would be SIA and the other is something like '4of7'


Most though are contactid by default and that is all some can do. It  
seems to pass more descriptive info of the alrams vs the other  
methods but the others do work.



On Dec 17, 2005, at 10:31 AM, Rich Adamson wrote:

I am trying to get some feedback from anyone who may have  
experience of

a problem I am having.
We have several buildings that having only fiber to them so in order
that the alarm panel can call the central station, I have provided a
Sipura 1001 ATA. I can make a call to the central station through the
ATA, using an analogue phone.
The alarm panel does not seem to function properly thorugh the ATA,
either it is not going off-hook properly or the ATA is treating the
modem tones as a fax, I am not sure what is happening. Does anyone  
have

experience of getting this to work?


Is the alarm panel truly expecting to use a modem to communicate with
central station?

If so, sip/rtp will not handle modems that attempt to use anything
greater then about 2400 baud. (Note: there is a diffence is the term
baud verses bits/second. Newer high-speed modems use an encoding
mechansim that involves phase-shift technology to achive a higher
bit-per-second speed over a low baud rate. sip/rtp will not accurately
reproduce any modem signal that involves phase-shifting. The sampling
rate is not sufficient to accurately reproduce phase-shifted analog
signals.)

Also, in the sipura release notes (for v3.1.5) specifically
indicates they watch for fax tones and, more recently, modem tones.

.Distinguish between FAX Passthrough mode and Modem Passthrough  
Mode.

 Modem Passthrough Mode can only be triggered by predialing the
 Modem Line Toggle Code. FAX Passthrough Mode is triggered by
 CED/CNG tone or NSE events. Echo canceller is automatically  
disabled
 for Modem Passthrough Mode only. Echo canceller is  
automatically disabled
 only if FAX Disable ECAN (Line 1/2) is set to yes for that  
line

 (in that case FAX passthrough is the same as Modem passthrough).
 Call-waiting and silence suppression is automatically disabled  
for
 both FAX and Modem passthrough as before. In addition, out-of- 
band
 DTMF Tx is disabled during modem or fax passthrough (all audio  
are


The Modem Line Toggle Code: option appears on the Regional tab and
has a default string of *99. Therefore, your alarm panel modem would
need to prefix its dialing with *99.

Might check those two items.

Rich


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RE: [Asterisk-Users] CID lookup from an Exchange Public folder

[Colin Anderson]

Exchange contacts != AD entries. Contacts in Exchange are basically email
messages with metadata.  Now, if all of your contacts WERE in AD, you could
do a script to query AD through LDAP (that's what AD is - LDAP with MS
extensions) and you would solve latency problems when Asterisk would query
AD instead of clunky MAPI. Here's a cool script to export contacts in a
public folder to AD:

http://www.msexchange.org/articles/Migrating-Contacts-Distribution-Lists-Out
look-Active-Directory.html

The problem with this is maintenance, since now you have 2 contact
databases. Making sure they are sync'd wouldn't be an automatic process and
invariably would mean that an admin would have to fire up ADSI Edit every
once in a while. This is mitigated by how often you change contacts. In an
org where contacts change rarely, or never this isn't a problem. Where I
work, contacts nmber in the THOUSANDS and change EVERY DAY. The
administrative overhead of maintaining those guys in AD is brutal, and
that's why at my work I have basically banned using public folders as a
contact manager and insisted that we use SQL server with a web front-end,
this makes things simple for the maintainer, extensible and fast, and SQL
server plugs into everything.

hth



-Original Message-
From: Steve Hanselman [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 17, 2005 3:18 AM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] CID lookup from an Exchange Public folder


We have a public folder full of contacts, but I understood that you
could only access this if the contacts were contacts in AD?

I was planning on doing a match on telephone number, mobile number and
fax.  And then pulling a shortened version of the name as the caller ID,

Steve


-Original Message-
From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 16 December 2005 21:48
To: asterisk-users@lists.digium.com
Cc: Steve Hanselman
Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder

Steve,

You can get to anything in Exchange via LDAP. What is and or is not
working? Where are you entering the callerID info you want pulled?
Please see attachment for where you might want to enter this. Please
share if you get this.

Cheers.

Jason


-
Message: 1
Date: Fri, 16 Dec 2005 18:29:12 -
From: Steve Hanselman [EMAIL PROTECTED]
Subject: [Asterisk-Users] CID lookup from an Exchange Public folder
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Has anybody done this?

I looked at LDAP but you can't get to them that way, I'm considering
either a timed export, or some other way (can you access them via IMAP?
Or by wget on the owa web structure?)

Steve

 


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Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UK
Registered Office as above. Registered in England No. 2764339

See our current vacancies at www.brendata.co.uk
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Re: [Asterisk-Users] Asterisk 1.2.1

[Tzafrir Cohen]

On Tue, Dec 13, 2005 at 04:57:08PM -0500, Leah Newmark wrote:
 Hi, All.
 
 We recently installed Asterisk 1.2.1 through the Debian package/CVS.

Are those self-made packages or packages from Sid? What do you mean by
CVS?

If official packages, I suggest you reportbug(1) . 

 
 The CLI, however, seems to be missing some of the commands I'm familiar
 with in older versions of Asterisk, namely the SIP and IAX2 commands, as
 well as extensions reload.
 
 I've viewed the changelog, and frankly I'm stumped to where they could
 have gone.
 
 Any help would me much appreciated!
 
 Thanks,
 
 Leah Newmark
 Capalon VoIP
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Re: [Asterisk-Users] Patch zaptel.init to support debian

[Tzafrir Cohen]

On Tue, Dec 13, 2005 at 05:21:42PM +, Karl O. Pinc wrote:
 
 On 12/13/2005 07:32:10 AM, Kevin P. Fleming wrote:
 
 This script is completely unnecessary on Debian; just add the modules  
 you wish to load into /etc/modules and they will be loaded at boot  
 time.
 
 FYI the list.  Using debian with linux 2.6 you don't do anything,
 the requsite module information is installed in /etc/modprobe.d/zaptel
 and it just works.

Allow me to correct this:

Here is something frm an Asterisk 1.0 system. It has not radically
changedin Asterisk 1.2:

# chat that specific file for the format...
$ grep e159 /lib/modules/`uname -r`/modules.pcimap
wct1xxp  0xe159 0x0001 0x6159 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa159 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xe159 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xb100 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xb1d9 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xb119 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa9fd 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa8fd 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa800 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa801 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa908 0x 0x 
0x 0x0
wcfxs0xe159 0x0001 0xa901 0x 0x 
0x 0x0
wcfxo0xe159 0x0001 0x8085 0x 0x 
0x 0x0
wcfxo0xe159 0x0001 0x8086 0x 0x 
0x 0x0
wcfxo0xe159 0x0001 0x8087 0x 0x 
0x 0x0
wcte11xp 0xe159 0x0001 0x71fe 0x 0x 
0x 0x0
wcte11xp 0xe159 0x0001 0x79fe 0x 0x 
0x 0x0
wcte11xp 0xe159 0x0001 0x795e 0x 0x 
0x 0x0
wcte11xp 0xe159 0x0001 0x79de 0x 0x 
0x 0x0
wcte11xp 0xe159 0x0001 0x797e 0x 0x 
0x 0x0
hisax0xe159 0x0002 0x 0x 0x 
0x 0x0
hisax0xe159 0x0001 0x 0x 0x 
0x 0x0

This information is generated by depmod even in 2.4 . It is extracted
from the declerations inside kernel modules code.

discover and kudzu use that information to load the relevant modules
when they are run. They scan the bus in a generally predictable order at
boot time. They don't require any special 2.6 features. But they don't
support hotplugging.

hotplug takes this to th next level: the kernel notifies userspace of
new events. This allows hotplugging.

The default configuration of zaptel from the tarball is to try to run
ztcfg (with too many v flags, mind you) after every zaptel module gets
loaded (even ztdummy, ztdynamic and zaptel, IIRC). As a result, modprobe
will fail many times due to ztcfg being run with an incomplete
configuration. 

The solution to that is to have a simple init.d script that modpobes
ztdummy if no other span is discovered and then runs ztcfg. As it has no
problems running twice, it needs no lock file.

Regarding the stop action: There is no need to rmmod those modules at
shutdown time, IMHO. Hence removing them should not be the default stop
action, should this be an init.d script.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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Re: [Asterisk-Users] Best way to automatically stop and start Asterisk nightly

[Tzafrir Cohen]

On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote:
 Hi,
 
 I am planning on restarting asterisk nightly as I seem to be 
 experiencing some sort of memory leak (Asterisk slows down over time). 

This is not an indication of a memory leak. The size of the asterisk
process:

  ps `cat /var/run/asterisk/asterisk.pid` -o vsize -o rss

Do those inflate over time?

 I 
 have reviewed the Asterisk suggestions for management and one item is 
 the routine rebooting of Asterisk.  Since I have Asterisk 1.2.1 what is 
 the recommend way to implement an automatic stop and start of asterisk 

asterisk -rx 'restart now' from a daily cron job?

Mind you, this is a bad patch and *NOT A FIX*.

 (there are changes in 1.2 with reload and restart) and is this enough or 
 should I restart the hardware as well??

If you suspect a user-space memory leak than restarting the application
should free that memory.

BTW: what do you mean by slow down?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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Re: [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk

[Tzafrir Cohen]

On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote:
 screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run 
 safe_asterisk in production

Any reason you need to run asterisk in a console?

asterisk -r allows you to view the current console.
/var/log/asterisk/messages gives you the full history: grep it, tail -f,
or whatever.

If it doesn't: configure /etc/asterisk/logger.conf to log what you want
to whereever you want.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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Re: [Asterisk-Users] Patch zaptel.init to support debian

[Tzafrir Cohen]

On Tue, Dec 13, 2005 at 06:26:49AM +, Karl O. Pinc wrote:
 Hi,
 
 Don't know if this is really right, all I know is
 that Debian sarge does not have /var/lock/subsys/.
 
 I foolishly made this patch against the zaptel 1.2
 branch rather than trunk, although I did check that
 the trunk has the problem.  It'll probably apply
 
 I ran it and it works for me.

It is not needed anyway for loading the modules on Debian, as modules
will be loaded before it by discover and/or hotplug (except the bristuff
modules).

So for the normal case, it is not used anyway. It is only used if you
ever want to rmmod those modules. For that you borrow can use genzaptelconf -u
from the latest Sid deb, which also has a nice safety check if asterisk is
running.

I generally prefer to load the modules in /etc/modules and remove the
insane post-install commands from modules.conf/modprobe.conf which are
the causes of strange error messages and of the fact that you modprobe
zaptel separately.

BTW: some modules may provide a span (/proc/zaptel/n , not necessarly
/proc/zaptel/1) but not function as a timing device. In which case
you'll still need to modprobe ztdummy, right?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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[Asterisk-Users] Linksys PAP2 and Asterisk

[Jason \(WeatherServer\)]

I'm sure this question has been asked before but I can't seem to find any 
info on it.

Is there anything special that needs to be setup on the PAP2 side and the 
Asterisk side for the PAP2 to work on the asterisk server.

I've entered all the settings for my VoIP provider but all I get is
Registration State: Can't connect to login server
  on the PAP2

Thank you.



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Re: [Asterisk-Users] md 3200

[Tzafrir Cohen]

On Tue, Dec 13, 2005 at 03:24:33PM -0500, Vladimir Montealegre wrote:
 i have two cards md3200 buy they dont work is possible connect two single 
 phone lines with 2 cards x100 clone ??

Basically, just as you can connect two phones on the same line: not
together. In practice: think of a line with two answering machine: who
will answer first? 

Not to mention some other practical problems.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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Re: [Asterisk-Users] Patch zaptel.init to support debian

[Kevin P. Fleming]

Tzafrir Cohen wrote:


BTW: some modules may provide a span (/proc/zaptel/n , not necessarly
/proc/zaptel/1) but not function as a timing device. In which case
you'll still need to modprobe ztdummy, right?


That would be true, although none of the drivers in the Zaptel source 
distribution fall into that category.


This would all be much, much simpler if ztdummy would be loaded all the 
time and just not be used if any other driver provided a timing device.

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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Tzafrir Cohen]

On Mon, Dec 12, 2005 at 11:28:35AM -0800, Johnny Voice wrote:
 For my asterisk installation in my lab, I will install the 

RedHat

 Linux ES v4 
 distribution (with kernel 2.6) onto a Dell Power Edge 1650 with 
 ~16GB of Raid-1 hard disk space.

Not much. Asterisk on its own doesn't take much either. However what
else do you need to run on that system besides Asterisk?

Below are some recommendations that are hopefully better than nothing:


   Before installing Linux, what should I set the following disk partitions 
 to?:
   (root)/
   /boot

Keep /boot off the RAID? Otherwise, I see little point in a separate
/boot .

   swap
   /usr

Again, I see a little point in a separate /usr . Consider keeping a
separate /usr/local , though.

   /home

An asterisk system typically does not have users and need nt have a
separate /home

   /tmp

Enoughtemporary space. Though if you have really really much memory, it
can be a tmpfs.

   /var

Here sit:

* logs
* mail spool
* mail messages
* asterisk sounds
* asterisk recordings
etc. 

In other words: it should be big. In some cases you should even create
seperate subpartitions under it to prevent them from filling up the rest
of the partition (e.g: to limit the size of the recording, should you
like to).

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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[Asterisk-Users] Grandstream GXP-2000 Auto Answer

[William M. Sandiford]

Has anyone been 
successful getting Auto-Answer by Call-Info to work with the GXP 
2000

I have followed the 
suggestions in

http://www.voip-info.org/wiki/view/GXP-2000

Specifically I 
have:

1. Upgraded to 
1.0.1.13, which supposedly supports this feature
2. Set Allow 
Auto-Answer by Call-Info to YES in the GXP2000 config
3. Used, 
SIPAddHeader(Call Info: answer-after=0) in my dialplan prior to the Dial 
command.

Still the phone just 
rings, and doesn't auto-answer.

Any 
suggestions?

Thanks in advance, 

Bill
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[Asterisk-Users] MusicOnHold not working

[Jason Lixfeld]

Running Asterisk 1.2.1 on Suse 10.0 X86-64.

Tried to get mpg123 0.59r which came with the 1.2.1 dist running on  
this box, but all I get is poop:


as   -o decode_i586.o decode_i586.s
decode_i586.s: Assembler messages:
decode_i586.s:44: Error: suffix or operands invalid for `push'
decode_i586.s:45: Error: suffix or operands invalid for `push'
decode_i586.s:46: Error: suffix or operands invalid for `push'
decode_i586.s:47: Error: suffix or operands invalid for `push'
decode_i586.s:67: Error: suffix or operands invalid for `push'
decode_i586.s:70: Error: suffix or operands invalid for `push'
decode_i586.s:81: Error: suffix or operands invalid for `push'
decode_i586.s:83: Error: suffix or operands invalid for `push'
decode_i586.s:86: Error: suffix or operands invalid for `push'
decode_i586.s:161: Error: suffix or operands invalid for `pop'
decode_i586.s:211: Error: suffix or operands invalid for `pop'
decode_i586.s:296: Error: suffix or operands invalid for `pop'
decode_i586.s:315: Error: suffix or operands invalid for `pop'
decode_i586.s:316: Error: suffix or operands invalid for `pop'
decode_i586.s:317: Error: suffix or operands invalid for `pop'
decode_i586.s:318: Error: suffix or operands invalid for `pop'
make[3]: *** [decode_i586.o] Error 1
make[3]: Leaving directory `/usr/src/asterisk-1.2.1/mpg123-0.59r'
make[2]: *** [mpg123-make] Error 2
make[2]: Leaving directory `/usr/src/asterisk-1.2.1/mpg123-0.59r'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/src/asterisk-1.2.1/mpg123-0.59r'
make: *** [mpg123] Error 2

None of my research revealed any answers so I started looking at  
mpg123 alternatives.  At the top of http://www.voip-info.org/tiki- 
pagehistory.php?page=Asterisk+mpg123+faking+itdiff2=3 there is a  
note that says how 1.2 has solved the MPG123 issue.  What does that  
mean exactly?  What has 1.2 solved WRT mpg123, and how has it solved  
it?  I couldn't find the answer so I kept digging and came across a  
reference for format_mp3.  I compiled and installed format_mp3 from  
1.2.1-addons and modified my musiconhold.conf as per the link, and  
ensured the module was loaded by asterisk:


[default]
mode=files
directory=/var/lib/asterisk/moh-native
random=yes

asterisk*CLI show modules like format_mp3
Module  
Description  Use Count
format_mp3.so  MP3 format [Any rate but 8000hz mono  
opt 0

1 modules loaded
asterisk*CLI

I copied the default musiconhold mp3 files from mohmp3 to moh-native  
so I had something to play.  When I call in, dial up the MoH test  
extension, I see musiconhold being called, but it's stopped  
immediately after it's started.  Logs show nothing so I'm kinda lost.


asterisk*CLI
-- Executing Answer(SIP/7501-4dd0, ) in new stack
-- Executing MusicOnHold(SIP/7501-4dd0, ) in new stack
-- Started music on hold, class 'default', on channel 'SIP/ 
7501-4dd0'

-- Stopped music on hold on SIP/7501-4dd0
asterisk*CLI

Anyone have any ideas?


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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Michiel van Baak]

/home
 
 An asterisk system typically does not have users and need nt have a
 separate /home

I disagree here.
You have at least 1 user to remotaly login to the system to
do some work on it. Think config changes etc.
In case of unauthorized access (ppl stole your password or
whatever) you will be glad you have /home on a seperate
partition that is mounted noexec,nosuid,nodev

 
/tmp
 
 Enoughtemporary space. Though if you have really really much memory, it
 can be a tmpfs.

Same here as with /home. Although this will need some
scripting work for apt/up2date cause they run the installer
scripts from /tmp

 
/var
 
 Here sit:
 
 * logs
 * mail spool
 * mail messages
 * asterisk sounds
 * asterisk recordings
 etc. 
 
 In other words: it should be big. In some cases you should even create
 seperate subpartitions under it to prevent them from filling up the rest
 of the partition (e.g: to limit the size of the recording, should you
 like to).

Even better would be to use LVM for /var partitions.
That way you can easily add extra space to it without the
hassle of moving around data.


All this is just my tipstricks archive for server installs.
Feel free to trash it cause it's by no means 'the way to do
it' It just works for me.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] TDM01B answering issue

[chawki hammoud]

Hi: 
i have these configured in zapata.conf:

signalling=fxs_ks
context=incoming
channel = 1

and these in extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DeadAGI(astcc.agi)
exten = s,3,Hangup

[tele]
exten = _01XX,1,Dial,ZAP/1/${EXTEN}

for example when i try to dial [EMAIL PROTECTED] the call
is been answered when it starts ringing and not when
the other side pick up the phone, how can i configure
my zapata.conf to answer call when the another side
pick up the phone?



--- Rich Adamson [EMAIL PROTECTED] wrote:

 
  Is there any possible way to make TDM01B  answers
 when
  the other side pick up the phone ,and to prevent
 it
  from answering just when it starts ringing?
 
 Yes, if I understand your question properly.
 
 Suggest you post relavent parts of zapata.conf and
 extensions.conf
 that are associated with the fxo port of the TDM
 card. No one can
 help you unless you provide _some_ detail how you
 have this stuff
 configured.
 
 
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Re: [Asterisk-Users] Best way to automatically stop and start Asterisk nightly

[Chuck Bunn]

Hi,

Thanks for the input. I will try your suggestions. By slowing down the 
server takes longer and longer to respond to prompts such as retrieving 
voice mail. I am recompiling my install this weekend as I have had a 
continued problem with logs (see other post) and this might be related 
to the problem. I will use your command to see if 'asterisk.pid' 
inflates over time...


Thanks Again

Tzafrir Cohen wrote:


On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote:
 


Hi,

I am planning on restarting asterisk nightly as I seem to be 
experiencing some sort of memory leak (Asterisk slows down over time). 
   



This is not an indication of a memory leak. The size of the asterisk
process:

 ps `cat /var/run/asterisk/asterisk.pid` -o vsize -o rss

Do those inflate over time?

 

I 
have reviewed the Asterisk suggestions for management and one item is 
the routine rebooting of Asterisk.  Since I have Asterisk 1.2.1 what is 
the recommend way to implement an automatic stop and start of asterisk 
   



asterisk -rx 'restart now' from a daily cron job?

Mind you, this is a bad patch and *NOT A FIX*.

 

(there are changes in 1.2 with reload and restart) and is this enough or 
should I restart the hardware as well??
   



If you suspect a user-space memory leak than restarting the application
should free that memory.

BTW: what do you mean by slow down?

 



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[Asterisk-Users] placing a call in one or several call groups

[hgaillac-sip]

Hello,
I read 
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups

So i set callgroup and pickupgroup in sip.conf .
How can I forward an incoming call to one or more
callgroup.

Regards 
Harry 






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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Andrew Kohlsmith]

On Saturday 17 December 2005 15:18, Michiel van Baak wrote:
 I disagree here.
 You have at least 1 user to remotaly login to the system to
 do some work on it. Think config changes etc.
 In case of unauthorized access (ppl stole your password or
 whatever) you will be glad you have /home on a seperate
 partition that is mounted noexec,nosuid,nodev

And I disagree with you.  :-)  My Asterisk installs are minimal.  Two 
partitions, one for / and one for /var, with /tmp symlinked to /var/tmp.  I 
have only two accounts log in, root and a script account, both using DSA 
keys.  I imagine you could put /home in /var/home but really it's not that 
critical for me.  If someone gains root or the script user access they can 
cause a lot more damage than any rootkit.

 Even better would be to use LVM for /var partitions.
 That way you can easily add extra space to it without the
 hassle of moving around data.

I use LVM for everything but /.  :-)

Good tips for general multiuser setups but I dunno; you can secure everything 
out the wazoo and just end up with a local root exploit crashing through all 
your security.  I prefer the minimal approach which doesn't let / fill up and 
if someone manages to grab a password... well you're screwed anyway.  
minimize the impact to other systems.  :-)

-A.

-A.
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Re: [Asterisk-Users] TDM01B answering issue

[Andrew Kohlsmith]

On Saturday 17 December 2005 15:23, chawki hammoud wrote:
 [tele]
 exten = _01XX,1,Dial,ZAP/1/${EXTEN}

 for example when i try to dial [EMAIL PROTECTED] the call
 is been answered when it starts ringing and not when

No, the call is *not* answered when you hit this line in the dialplan.  If 
this is occuring you have other issues that your paste did not reveal.

Where are you Dialing [EMAIL PROTECTED] from?  Is it from some other system 
that 
has an Answer() in its dialplan?  Or from an FXS port that has a 
immediate=yes set?

-A.
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[Asterisk-Users] Sangoma E1 board Experience

[[EMAIL PROTECTED]]

hi,

Do anyone have experience with the Sangoma E1 A102 or A104 etc?

I am tempted to buy one for testing out, but I don't want to waste more 
money and find that they have the same issues as the Digium's.


I know Sangoma have a better solution to IRQ problems, but I know 
nothing about their PCI interface. Do they have a i/o buffer that allow 
more than 8 bytes to be buffered per channel etc?


Any experience on this?

jvb
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Re: [Asterisk-Users] Sangoma E1 board Experience

[Andrew Kohlsmith]

On Saturday 17 December 2005 16:21, [EMAIL PROTECTED] wrote:
 I am tempted to buy one for testing out, but I don't want to waste more
 money and find that they have the same issues as the Digium's.

They work about the same.  I've never had IRQ issues with Digium though (even 
sharing IRQs).

 I know Sangoma have a better solution to IRQ problems, but I know
 nothing about their PCI interface. Do they have a i/o buffer that allow
 more than 8 bytes to be buffered per channel etc?

No I do not believe so.  Zaptel's pretty strict about keeping the amount of 
queued data to an absolute minimum.

-A.
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Re: [Asterisk-Users] Sangoma E1 board Experience

[[EMAIL PROTECTED]]

No I do not believe so.  Zaptel's pretty strict about keeping the amount of 
queued data to an absolute minimum.
 


Do you know if this is a driver or hardware limitation?

Jan
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Re: [Asterisk-Users] TDM01B answering issue

[chawki hammoud]

HI:
I dial this on console :
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp, zap/1/01472345) in
new stack
-- Called 1/01472345
-- Zap/1-1 answered OSS/dsp
  Console call has been answered 

the call here to 01472345 is been answered before the
other side (01472345 side) pick up the phone .Do you
have any clue? 




--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:

 On Saturday 17 December 2005 15:23, chawki hammoud
 wrote:
  [tele]
  exten = _01XX,1,Dial,ZAP/1/${EXTEN}
 
  for example when i try to dial [EMAIL PROTECTED] the
 call
  is been answered when it starts ringing and not
 when
 
 No, the call is *not* answered when you hit this
 line in the dialplan.  If 
 this is occuring you have other issues that your
 paste did not reveal.
 
 Where are you Dialing [EMAIL PROTECTED] from?  Is it
 from some other system that 
 has an Answer() in its dialplan?  Or from an FXS
 port that has a 
 immediate=yes set?
 
 -A.
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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Tzafrir Cohen]

On Sat, Dec 17, 2005 at 09:18:39PM +0100, Michiel van Baak wrote:
 /home
  
  An asterisk system typically does not have users and need nt have a
  separate /home
 
 I disagree here.
 You have at least 1 user to remotaly login to the system to
 do some work on it. Think config changes etc.
 In case of unauthorized access (ppl stole your password or
 whatever) you will be glad you have /home on a seperate
 partition that is mounted noexec,nosuid,nodev

noexec? What will that give you against a user with a shell acount?

[EMAIL PROTECTED]:~/Proj/Debs/Netcat/netcat-1.10$ 
$ cp /bin/ech /tmp/echonoexec
$ chmod 644 /tmp/echonoexec
$ ls -l /tmp/echonoexec
-rw-r--r--  1 tzafrir tzafrir 13912 2005-12-17 23:52 /tmp/echonoexec
$ /lib/ld-linux.so.2 /tmp/echonoexec it runs!
it runs!

Not to mention all of the #! executables. Only static executables are
harmed. So what was it that noexec prevented me form doing?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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Re: [Asterisk-Users] Linksys PAP2 and Asterisk

[John Biundo]

Hi Jason,

I've got several PAP2s working with asterisk.  Feel free to e-mail me 
off-line if you want to compare configurations.  Which version of 
asterisk and which PAP2 firmware are you running?


Cheers,
john
Jason (WeatherServer) wrote:
I'm sure this question has been asked before but I can't seem to find any 
info on it.


Is there anything special that needs to be setup on the PAP2 side and the 
Asterisk side for the PAP2 to work on the asterisk server.


I've entered all the settings for my VoIP provider but all I get is
Registration State: Can't connect to login server
  on the PAP2

Thank you.



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[Asterisk-Users] Can't pickup call when dialing *8 extension

[hgaillac-sip]

Hello,

I added callgroup=1 and pickupgroup=1 for sip channels
however I can't pickup a call (see below ) between sip
phones when i dial *8 .

May I have to add app_pickup to solve this problem.
I use asterisk-1.2

Regards
Harry


serveur1*CLI
-- SIP read from 80.119.8.167:5060:
ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
From: alice sip:[EMAIL PROTECTED];tag=AF3B88E-55239161
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as543ba455
CSeq: 2 ACK
User-Agent: Sip EXpress router(0.9.4 (i386/linux))
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call
'[EMAIL PROTECTED]'
-- Nobody picked up in 1 ms
Reliably Transmitting (NAT) to 80.119.8.167:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167:5050;branch=z9hG4bK60e70916;rport
From: alice
sip:[EMAIL PROTECTED]:5050;tag=as7cefba23
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5050
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0







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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Michiel van Baak]

On 00:03, Sun 18 Dec 05, Tzafrir Cohen wrote:
 On Sat, Dec 17, 2005 at 09:18:39PM +0100, Michiel van Baak wrote:
  /home
   
   An asterisk system typically does not have users and need nt have a
   separate /home
  
  I disagree here.
  You have at least 1 user to remotaly login to the system to
  do some work on it. Think config changes etc.
  In case of unauthorized access (ppl stole your password or
  whatever) you will be glad you have /home on a seperate
  partition that is mounted noexec,nosuid,nodev
 
 noexec? What will that give you against a user with a shell acount?
 
 [EMAIL PROTECTED]:~/Proj/Debs/Netcat/netcat-1.10$ 
 $ cp /bin/ech /tmp/echonoexec
 $ chmod 644 /tmp/echonoexec
 $ ls -l /tmp/echonoexec
 -rw-r--r--  1 tzafrir tzafrir 13912 2005-12-17 23:52 /tmp/echonoexec
 $ /lib/ld-linux.so.2 /tmp/echonoexec it runs!
 it runs!
 
 Not to mention all of the #! executables. Only static executables are
 harmed. So what was it that noexec prevented me form doing?

I agree with this.
But noexec is not the only thing.
As this was not really a security thread, I just posted my
personal prefs.
Together with those mount options I also use systrace.
There I disable the /lib/ld-linux hacks and stuff.

Like I said, my setup is not the way to do it.
It's just what works for me.
I was commenting on the fact ppl think having seperate
partitions for different parts of a system is not what is
needed. There are some uses for it, that's what it was all
about.
Having partitions with mount options is not the only step in
securing your system, that much is shown here ;)

Actually in my setup my /home is not even local. That is
just another reason to setup a box with seperate partitions
for /home, /tmp, /usr etc. It will save you time in the
occasion you want to deploy a remote filesystem for one of
them.

I'm sorry if you took my points as attacks on your setup.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] Linux Partitions (before asterisk install)

[Michiel van Baak]

On 15:41, Sat 17 Dec 05, Andrew Kohlsmith wrote:
 On Saturday 17 December 2005 15:18, Michiel van Baak wrote:
  I disagree here.
  You have at least 1 user to remotaly login to the system to
  do some work on it. Think config changes etc.
  In case of unauthorized access (ppl stole your password or
  whatever) you will be glad you have /home on a seperate
  partition that is mounted noexec,nosuid,nodev
 
 And I disagree with you.  :-)  My Asterisk installs are minimal.  Two 
 partitions, one for / and one for /var, with /tmp symlinked to /var/tmp.  I 
 have only two accounts log in, root and a script account, both using DSA 
 keys.  I imagine you could put /home in /var/home but really it's not that 
 critical for me.  If someone gains root or the script user access they can 
 cause a lot more damage than any rootkit.

true. No setup is secure. The only security is disconnecting
your system from the net ;)
 
  Even better would be to use LVM for /var partitions.
  That way you can easily add extra space to it without the
  hassle of moving around data.
 
 I use LVM for everything but /.  :-)

Same here. drbd devices as low-level with lvm on top of it.

 
 Good tips for general multiuser setups but I dunno; you can secure everything 
 out the wazoo and just end up with a local root exploit crashing through all 
 your security.  I prefer the minimal approach which doesn't let / fill up and 
 if someone manages to grab a password... well you're screwed anyway.  
 minimize the impact to other systems.  :-)

This is becoming a thread that totally looses track of the
OP question. Security is a complex issue and every
system/install needs it's own policy.
Like I said, I was just posting my own view on things.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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[Asterisk-Users] Teliax billing question

[Ryan Burke]

Teliax users,

I have a couple questions about Teliax, just 
hopeing some current customers might shed some light on them.

How reliable is a toll-free number from Teliax? Has 
anyone had any problems with it?

The Pay as you go plan has a Billing of 60/1, what 
does that mean? My guess is 60 seconds minimum (does this apply for incoming AND 
outgoing, or just outgoing) and a period of 1 bill per month?

For a total bill per month, it would be $.02 per 
outgoing minute (with a minimum of 60 seconds per call)and $.029/minute 
for incoming toll free (don't know if minimum time applies here) plus $4.99 for 
a toll-free number, right? Is there any other charges because of the toll free 
number?

Thanks for your help,
Ryan

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Re: [Asterisk-Users] Grandstream GXP-2000 Auto Answer

[Julian J. M.]

Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom

From that article:
There is an 'allpage.agi' now available at
http://aussievoip.com.au/allpage.agi. Documentation is available in
the file. This should work with Snom and Grandstream GXP2000 phones
(and possibly budgettones if they roll the changes across) with
firmware greater than 1.0.13 (not publically available at time of
writing, due out in October 2005)

I've used that with my GXP-2000, and seems to work ok. I had, however,
to adapt it to my needs.

Regards
Julian J. M.

On 12/17/05, William M. Sandiford [EMAIL PROTECTED] wrote:

 Has anyone been successful getting Auto-Answer by Call-Info to work with the
 GXP 2000

 I have followed the suggestions in

 http://www.voip-info.org/wiki/view/GXP-2000

 Specifically I have:

 1.  Upgraded to 1.0.1.13, which supposedly supports this feature
 2.  Set Allow Auto-Answer by Call-Info to YES in the GXP2000 config
 3.  Used, SIPAddHeader(Call Info: answer-after=0) in my dialplan prior to
 the Dial command.

 Still the phone just rings, and doesn't auto-answer.

 Any suggestions?

 Thanks in advance,
 Bill
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[Asterisk-Users] Re: Teliax billing question

[Wolfgang S. Rupprecht]

Ryan Burke [EMAIL PROTECTED] writes:
 Is there any other charges because of the toll free number?

I was toying with the idea of getting an 800 number too, but the issue
of a substantial per call fee for pay-phones calls has me worried.
Hopefully someone here can clarify what the deal is there.  I've seen
numbers quoted as high as a 60-cents for the payphone settlement.

from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html

Watch Out for New 1-800 Number Scam - An old scam may be cropping up
again for consumers with personal 1-800 numbers. Most long distance
companies charge subscribers a per-call fee for calls placed from a
payphone to a residential 1-800 number. This fee is then sent back to
the owner of the payphone. While this arrangement is perfectly
legitimate, in 2002, scammers in Berkeley, California found a way to
take advantage of the system. They set up a phony payphone company and
connect a bank of payphones to an automatic dialer. The dialer then
randomly dialed 1-800 numbers until it hit a residential toll-free
number. When the call is picked up, the scammer pocketed the 24¢
fee. Thanks to the auto-dialer, they could quickly rack up profits
from the scam. By the time the operation was shut down by police, they
had netted almost a half million dollars. Reports of a similar scam
are coming in and consumers with residential 800 numbers are urged to
check their April and May long distance bills for mysterious
one-minute phone calls from Denver, Colorado. If you find such a call,
be sure to contact your phone company. For more information on this
scam, click herei. (Thanks to ConsumerWorld.org for this tip.)

WIRELESS WATCH

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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[Asterisk-Users] Toll Free Providers

[Tom Vile]

Looking for a good toll free DID provider.  Any suggestions?

All ready tried Sellvoip and Gafachi and the experience was not desirable.

Thanks,

Tom Vile
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Re: [Asterisk-Users] Can't pickup call when dialing *8 extension

[Rich Adamson]

You might have to use *8#. At least I do with my 7960.



 I added callgroup=1 and pickupgroup=1 for sip channels
 however I can't pickup a call (see below ) between sip
 phones when i dial *8 .
 
 May I have to add app_pickup to solve this problem.
 I use asterisk-1.2
 
 Regards
 Harry
 
 
 serveur1*CLI
 -- SIP read from 80.119.8.167:5060:
 ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0
 Via: SIP/2.0/UDP
 80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
 From: alice sip:[EMAIL PROTECTED];tag=AF3B88E-55239161
 Call-ID: [EMAIL PROTECTED]
 To: sip:[EMAIL PROTECTED];tag=as543ba455
 CSeq: 2 ACK
 User-Agent: Sip EXpress router(0.9.4 (i386/linux))
 Content-Length: 0
 
 --- (8 headers 0 lines)---
 Destroying call
 '[EMAIL PROTECTED]'
 -- Nobody picked up in 1 ms
 Reliably Transmitting (NAT) to 80.119.8.167:5060:
 CANCEL sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP
 80.119.8.167:5050;branch=z9hG4bK60e70916;rport
 From: alice
 sip:[EMAIL PROTECTED]:5050;tag=as7cefba23
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]:5050
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 CANCEL
 User-Agent: Asterisk PBX
 Max-Forwards: 70
 Content-Length: 0
 
 
 
   
 
   
   
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---End of Original Message-


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Re: [Asterisk-Users] Re: Teliax billing question

[Ryan Burke]

wolfgang,

Thanks for the heads up. I'm hoping to get some feedback from Teliax 
toll-free customers and see if they would recommend the service. Plus I have 
those few questions on billing.


Thanks again,
Ryan

- Original Message - 
From: Wolfgang S. Rupprecht [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Saturday, December 17, 2005 7:05 PM
Subject: [Asterisk-Users] Re: Teliax billing question



Ryan Burke [EMAIL PROTECTED] writes:

Is there any other charges because of the toll free number?


I was toying with the idea of getting an 800 number too, but the issue
of a substantial per call fee for pay-phones calls has me worried.
Hopefully someone here can clarify what the deal is there.  I've seen
numbers quoted as high as a 60-cents for the payphone settlement.

   from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html

   Watch Out for New 1-800 Number Scam - An old scam may be cropping up
   again for consumers with personal 1-800 numbers. Most long distance
   companies charge subscribers a per-call fee for calls placed from a
   payphone to a residential 1-800 number. This fee is then sent back to
   the owner of the payphone. While this arrangement is perfectly
   legitimate, in 2002, scammers in Berkeley, California found a way to
   take advantage of the system. They set up a phony payphone company and
   connect a bank of payphones to an automatic dialer. The dialer then
   randomly dialed 1-800 numbers until it hit a residential toll-free
   number. When the call is picked up, the scammer pocketed the 24¢
   fee. Thanks to the auto-dialer, they could quickly rack up profits
   from the scam. By the time the operation was shut down by police, they
   had netted almost a half million dollars. Reports of a similar scam
   are coming in and consumers with residential 800 numbers are urged to
   check their April and May long distance bills for mysterious
   one-minute phone calls from Denver, Colorado. If you find such a call,
   be sure to contact your phone company. For more information on this
   scam, click herei. (Thanks to ConsumerWorld.org for this tip.)

   WIRELESS WATCH

-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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[Asterisk-Users] SIP and echo cancel

[Mike Bernson]

I known that sip channel should be free from echo. I am find this is not the case for me.

The setup here is
Sipura 3000 connected to vonage
extensions are SIPURA 841 or SIPURA 2002 ATA.

I am getting echos on some of the outbound calls. I would like to be able to have one
of the software echo cancel working on SIP channel (RTP stream)

Before I start hacking this into asterisk 1.2.1 I would like to known if others are running
into this kind of problem ?

Is anyone working on this problem ?

If I make this work what is needed to get it put into a standard release ?


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Re: [Asterisk-Users] Teliax billing question

[Rich Adamson]

 I have a couple questions about Teliax, just hopeing some current customers 
 might 
shed some light on them.
  
 How reliable is a toll-free number from Teliax? Has anyone had any problems 
 with it?

They have been very reliable for me. Once in a great while they'll have
a problem, but then every company does.

 The Pay as you go plan has a Billing of 60/1, what does that mean? My guess 
 is 60 
seconds minimum (does this apply for
 incoming AND outgoing, or just outgoing) and a period of 1 bill per month?

That sounds right. Calls are rounded up to the next whole minute. Billing is
once a month.

 For a total bill per month, it would be $.02 per outgoing minute (with a 
 minimum of 
60 seconds per call) and $.029/minute for
 incoming toll free (don't know if minimum time applies here) plus $4.99 for a 
toll-free number, right? 

Yes, that's right.

 Is there any other charges because of the toll free number?

Nope. (I haven't paid any attention, but I don't think there are any
taxes on it either.)

To say all of the above a little clearer all usage (regardless of
incoming or outgoing) is $0.02/min, except for incoming 800 calls that
are currently invoiced at $0.029/min. Then add $4.99/mo for each 800
or DID number, and that's it. 

The pay as you go plan also supports an unlimited number (within reason)
of simultanous calls, so busy tone is essentially the result of how
you program your asterisk.

A rather nice feature is you give a credit card number and they will
automatically replenish/invoice you when needed, and you set the amount
on their web site as to how much you want them to replenish your account.
So, if you plan on low usage and replenish the account with $50.00, then
invoicing really occurs whenever your account runs out of funds, regardless
of whether its every two weeks or once in six months.

Quality of audio is very good, you get to choose the codec you want to 
use, and you choose the CallerID that you want for outgoing calls.

After you sign up for their service you will be given a login userid and
password to access your account info. That web site leaves a little to be
desired in terms of clearity, as some configuration options are not very
clearly identified. But, once you know that, then its not difficult to
adjust your callerid, sip vs iax preference, codec preference, etc.

Rich


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Re: [Asterisk-Users] Re: Teliax billing question

[Rich Adamson]

  Is there any other charges because of the toll free number?
 
 I was toying with the idea of getting an 800 number too, but the issue
 of a substantial per call fee for pay-phones calls has me worried.
 Hopefully someone here can clarify what the deal is there.  I've seen
 numbers quoted as high as a 60-cents for the payphone settlement.
 
 from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html

The scam isn't new, and its certainly not limited to home 800 numbers.
The same basic principles were used by some of the 900 number folks
a few years ago as well.

Hell, even the local telephone companies are doing weird billings, and
the average home owner never bothers to read the details to even
recognize it. 

It's really no different then any other invoice; pay attention to what
you are being invoiced.


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Re: [Asterisk-Users] Sangoma E1 board Experience

[Rich Adamson]

 
 No I do not believe so.  Zaptel's pretty strict about keeping the amount of 
 queued data to an absolute minimum.
   
 
 Do you know if this is a driver or hardware limitation?

Its not a limitation. Its an architectural design which is based on pulse 
code modulation (pcm) standards, which essentially says:
 - 8,000 audio samples per second,
 - each sample is an 8-bit value
 - resulting in 64,000 bits/second (like g711 codec standard)


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[Asterisk-Users] aastra.cfg mac.cfg examples Firmware version 1.3

[Lists]

I have gotten the tftp server working and the 9133i is doing a firmware
update and finds the aastra.cfg file as well as the 00XXX.mac file.  The
issue is that I can't figure out what is wrong in the configuration files
that it is not loading the extension, proxy, etc. info.

Could someone post their aastra.cfg file and mac.cfg file for a 9133i and/or
480i phone as well?

I would like to see a working copy of each and compare it to what I am doing
wrong.

Thanks.




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RE: [Asterisk-Users] Grandstream GXP-2000 Auto Answer

[Rob Thomas]

 Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
 
 From that article:
 There is an 'allpage.agi' now available at
 http://aussievoip.com.au/allpage.agi. Documentation is available in

I'm the author of that, and I've actually re-written it, because I was
pretty unhappy with the way it checked for in-use devices, etc.

I'll update the wiki and post the patched one later today, hopefully.

--Rob

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Re: [Asterisk-Users] TDM01B answering issue

[Eric \ManxPower\ Wieling]

*sigh*  Analog Zap FXO ports consider the call answered as soon as 
it's finished throwing the DTMF at the telco.  This is because a Zap 
port CAN'T tell when an analog call has been answered.


Andrew Kohlsmith wrote:

On Saturday 17 December 2005 15:23, chawki hammoud wrote:

[tele]
exten = _01XX,1,Dial,ZAP/1/${EXTEN}

for example when i try to dial [EMAIL PROTECTED] the call
is been answered when it starts ringing and not when


No, the call is *not* answered when you hit this line in the dialplan.  If 
this is occuring you have other issues that your paste did not reveal.


Where are you Dialing [EMAIL PROTECTED] from?  Is it from some other system that 
has an Answer() in its dialplan?  Or from an FXS port that has a 
immediate=yes set?


-A.
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Re: [Asterisk-Users] TDM01B answering issue

[Andrew Kohlsmith]

On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote:
 *sigh*  Analog Zap FXO ports consider the call answered as soon as
 it's finished throwing the DTMF at the telco.  This is because a Zap
 port CAN'T tell when an analog call has been answered.

Bah, you're absolutely correct.  I keep forgetting about POTS; I think PRI 
when I think Zap.

-A.
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Re: [Asterisk-Users] Re: Teliax billing question

[Ryan Burke]

Rich,

Thanks for your feedback. Sounds like what I was looking for. I think I'll 
sign up tonight!


Thanks,
Ryan


- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Saturday, December 17, 2005 8:30 PM
Subject: Re: [Asterisk-Users] Re: Teliax billing question





 Is there any other charges because of the toll free number?

I was toying with the idea of getting an 800 number too, but the issue
of a substantial per call fee for pay-phones calls has me worried.
Hopefully someone here can clarify what the deal is there.  I've seen
numbers quoted as high as a 60-cents for the payphone settlement.

from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html


The scam isn't new, and its certainly not limited to home 800 numbers.
The same basic principles were used by some of the 900 number folks
a few years ago as well.

Hell, even the local telephone companies are doing weird billings, and
the average home owner never bothers to read the details to even
recognize it.

It's really no different then any other invoice; pay attention to what
you are being invoiced.


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Re: [Asterisk-Users] TDM01B answering issue

[Steve Underwood]

Andrew Kohlsmith wrote:


On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote:
 


*sigh*  Analog Zap FXO ports consider the call answered as soon as
it's finished throwing the DTMF at the telco.  This is because a Zap
port CAN'T tell when an analog call has been answered.
   



Bah, you're absolutely correct.  I keep forgetting about POTS; I think PRI 
when I think Zap.
 


Its not absolutely correct, but its relatively correct. :-)

The above is true for most analogue lines around the world. However, 
there are some places which provide a positive answer indication on 
analogue lines. The form varies, but it is typically a reversal of line 
power, or a short timed break in line power. Similarly, while most of 
the world's analogue lines no longer provide a positive indication of 
hangup, some still do. Again, this is usually by reversal or a short 
timed break.


Steve

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Re: [Asterisk-Users] hint on Zap channels

[C F]

On 12/16/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
 C F wrote:
  Kevin, I'm not sure this would work here, but maybe it would.
  There was a bug posted about being able to use hint against local
  channels, would that not help him?
 
  http://bugs.digium.com/view.php?id=5779nbn=4

 No, the issue is that multiple ISDN devices are not distinct channels as
 far as Asterisk is concerned; they are all 'Zap/1' with different
 extensions behind that channel.

Kevin, I understood that. However, if that patch allows to create a
hint for the local channel no matter what extension that local channel
dials (which I'm not sure it does), then it is possilbe to do just
that, by creating hints for those extensions using the local channel,
and always dial those extensions only using the local channel.


 This is the same question as asking 'if I have a PRI connected to my
 Panasonic PBX, can I use hints for all the extensions on that PBX'. It
 won't work in Asterisk, because it's not aware of the actual endpoints,
 only the channel that connects to them.

Well, with my idea above it will, since you are dialing it using a
local channel. On inbound calls - from that other PBX to asterisk - it
could also be done with some DP magic, if that bug supports hints on
the local channels.

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Re: [Asterisk-Users] HW Echo Cancellers

[C F]

On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
 $1k for a single port T1


 I've gone down the Tellabs route, and am infinitely more happy.thanks C F 
 for the docs..

 -D


NP, anytime :) You seen the pics of the 253C? it's on the wiki. I'm
still looking for detailed docs on that. But I think I'll manage with
just what I posted there.

Anyhow, I called Orion and they told me $900.00 (US) for the single T1
desktop model.


 

 From: [EMAIL PROTECTED] on behalf of Steve Davies
 Sent: Fri 12/16/2005 10:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] HW Echo Cancellers



 On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
  I have used the orion...you can buy right from them.  However, I was not
  impressed with their sales teamI have one on a beta test, and they
  threatened to call a collection agency in when I refused paybent before
  the beta expired.
 

 Can you give an indication of price for their units? I've tried
 mailing a couple of times, but received no answer. I am just
 interested to know what price range we'd be looking at.

 Thanks,
 Steve
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Re: [Asterisk-Users] HW Echo Cancellers

[C F]

On 12/16/05, Steve Davies [EMAIL PROTECTED] wrote:
 On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote:
  $1k for a single port T1
 
  I've gone down the Tellabs route, and am infinitely more happy.thanks C 
  F for the docs..
 

 Tellabs looks a little too up-scale for what I need :). $1k for a
 single port orion unit might be worth considering for really stubborn
 installs though.


Why? they go for around $100.00 on eBay.

 Does anyone else have suggestions for external E1 hardware echo
 canceller solutions? In my case, I would be interested in one or two
 port desktop offerings, but I'm interested in larger scale for
 research purposes.

 Thanks again.
 Steve
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[Asterisk-Users] indications.conf for Japan?

[Robert La Ferla]

Anyone have an indications.conf entry for Japan?

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Re: [Asterisk-Users] Mediatrix 1204 help please.

[C F]

On 12/16/05, Rich Adamson [EMAIL PROTECTED] wrote:

  OK we need some help in setting up a good wiki-info page for setting up the 
  Mediatrix
 1204 to work with asterisk.  If anyone has
  set these unit's up and have them working please post your settings here so 
  we can
 create a page on the wiki. These unit's are
  being sold to be used via sip format with asterisk and there is no real 
  information
 on getting them working.  At present there one
  of the worst I have run into to get correctly working. These are very 
  expensive and
 some of us can't afford to send them back for
  a restocking fee.
 
  If someone working with Mediatrix has a white paper on getting these unit's 
  working
 please let us know the link for it.  It would
  be very helpful for many asterisk users.
 

 If you search the -users archives, you'll see where a couple of people
 have made them work. I believe there was at least one posting reflecting
 a working config.

 I did an eval on the 1204 in early 2004, but did not care for the way
 it interfaced with asterisk. The 1204 was really intended to interoperate
 with the 1104 as a toll bypass box.

 I was able to make it work and the audio was excellent with no echo
 whatsoever. Key items (in early 2004) included:
 - the 1204 does not have any sip register functions. One must configure it
   (and asterisk) to work with static IP addresses (instead of relying on
   the registration process).
 - calls from asterisk to (or through) the 1204 are treated as a group
   and the 1204 chooses the first available pstn port for all calls. If you
   want to direct a call to a specific port, one has to jump through hoops
   to force a CallerID (from asterisk) and then program 1204 to look for the
   callerid (which is then used to match a port number). Not cool.
 - programming the 1204 could only be done via snmp, and the snmp facility
   provided only ran on Windows. Each firmware upgrade to the 1204 required
   a new snmp implementation as the mib variables constantly changed. The
   snmp community string (eg, password) could not be changed from public,
   therefore exposing the 1204 to the internet would be a major security
   risk. (If you know snmp extemely well, you can use the mib definitions
   within a linux system to program the box, but you better be very good
   at snmp to do that.)

They now support a limited set of things that could be programmed via http.

 - Support for the box is only offered through resellers, and their typical
   resellers are those firms reselling traditional pbx's. A fair number of
   those don't have a clue what voip is about and even fewer can spell
   asterisk.

I thought they teach you how to spell asterisk in 4th grade math? :)

 - All firmware upgrades are chargable regardless of what problem might be
   found. The upgrade charge was very high (something like $500 in 2004).

 Given the above (in 2004), the risk associated with using the 1204 was
 far to great and I returned the unit for full credit. (The eval was arranged
 through Mediaxtix sales rep even though the unit came from a reseller.)

 I've not touched or seen the 1204 since early 2004, so can't help any more
 then what is stated above. The product may have improved since then, but
 I don't have clue what might have changed.



I have set it up many times now, but each time I have to do that, it
takes around an hour, which is a very long time for me compared to an
Adit or Sipura, but because of the quality of the box (sound is very
good) I still use it.
Each time I do it I just go thru the SNMP settings one by one, and
set/change what I think/know is needed, then test it to make sure it
works as I want. It would take me an extra hour to write it down and
post on the wiki, but I'll try to do it next time anyhow, that way it
might save me some time in the future as well :)
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[Asterisk-Users] ztdummy problem !!!

[Gabriel Sartor]

Hey, I´m trying to modprobe ztdummy, but when i make modprobe, return one error.

I use kernel 2.4 and have UHCI USB Controller allowed in my kernel.

This problem can be, because i dont have any pci card (fxo) at the computer ?
Thanks.
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Re: [Asterisk-Users] SIP and echo cancel

[Luki]

 Before I start hacking this into asterisk 1.2.1 I would like to known
 if others are running into this kind of problem ?

Asterisk doesn't do any echo cancellation in the setup you describe;
it just passes the audio data, and transcodes if necessary. The
endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible
for cancelling echo.

The Sipura ATA's generally do a good job cancelling echo. You may want
to play with the gain settings in the admin web config for the Sipura
ATA. As far as the 841 is concerned, if the handset volume is too loud
I noticed you may be getting acoustic echo. Hasn't been a problem for
me for PSTN calls or SIP to SIP calls though.

If you really want to patch asterisk to apply echo cancellation on the
RTP stream on pure VoIP calls, that would be interesting to see how
well it works.

--Luki
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Re: [Asterisk-Users] ztdummy problem !!!

[Luki]

 modprobe, return one error.
What is the error?

Ztdummy is an alternative if you don't have a hardware timing source,
so not having a PCI FXO card is not the cause.
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