RE: [Asterisk-Users] Dumping queue_log to MySQL

[Anton Krall]

Im using the fifo approach.. working great so 
far!

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kevin 
  SavoySent: Friday, May 05, 2006 8:57 AMTo: 'Asterisk 
  Users Mailing List - Non-Commercial Discussion'Subject: 
  [Asterisk-Users] Dumping queue_log to MySQL
  
  
  Anyone have a working solution for this? I played with 
  the demo that came with QueueMetrics to see how they were doing it and it was 
  working for a bit but now somehow every night it stopped. Perl and Tail are 
  still running on the server but the information is not dumping to the MySQL 
  database. I dont get any error messages anywhere telling me why it stops. As 
  far as tail and perl are concerned everything is fine. 
  We 
  will be using this for a call center and need more reliability. Anyone got one 
  working?
  
  Thanks
  
  _
  
  Kevin 
  Savoy
  Business Unit 
  Telecom Analyst
  2218 4th Ave W
  Williston, ND 58801
  Ph: 701-774-4023
  Fax: 701-774-2901
  http://www.novo1.com
  Novo 1 is a service mark of Novo 1, 
  Inc
  
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[Asterisk-Users] RE: Configure and Replace Voipjet.com content in Asterisk

[Crazy Boy]

Hi Chris,  Thank you for the response.For password information, its not a problem. Because, its trail version. So, we can register again with different mail id.  Can you please give me complete example. Here I am sending my config details and information provided by VoIPJet. So, please modify my config files according to the values provided by VoIPJet and send me.  Please assume 001-857-991-8585 as a phone number, I want to make a call.  My configuration details are:  Contents of IAX.CONF File: [102] type=friend username=102 secret=chandra host=dynamic context=tutorial  Contents of SIP.CONF File: [102] type=friend username=102 secret=chandra host=dynamic context=tutorial  Contents of EXTENSIONS.CONF File: [tutorial] exten = 102,1,Dial(SIP/102,10) exten = 102,2,Voicemail(u102) exten = 102,3,Voicemail(b102) exten = 102,4,Hangup  Contents of VOICEMAIL.CONF File: [default] 102 =
 chandra,Chandramouli,[EMAIL PROTECTED],[EMAIL PROTECTED] The above information is my content. With this Intercom is implemented successfully with voicemail.  The below information is given by VoIPJet:  VoipJet account number (username/UserID): 9333  Authorization code (password): a47769538c462223 (You should see an MD5 string, if it is blank logout and login again) Peer1 East Coast Server: 64.34.45.100 NAC East Coast Server II: 66.246.220.19 Mzima West Coast Server: 72.34.43.5 InterNap West Coast Server II (soon to be discontinued): 69.25.60.30 (Choose depending on your location) (N.B.
 216.118.117.46 has been discontinued due to problems)Asterisk PBX Step 1: Add the following lines to the end of iax.conf (found in /etc/asterisk)[voipjet]   type=peer   host= 64.34.45.100  secret= a47769538c462223  auth=md5   notransfer=yes   context=default  Step 2: Add the following to extensions.conf (found in /etc/asterisk)   ; NANPA: North American Numbers dialed as 1 + area code  ; For example, the New York Public Library is dialed as 12123400849  ; 1 (North American call) 212 (New York area code) 3400849 (libary's phone number)  ; WORLD: International Numbers dialed as
 011 + country code + number  ; For example, the Tate Modern Museum in London, U.K. is dialed as 011442078878000  ; 011 (International call) 44 (U.K. country code) 2078878000 (museum's number)  ; Finally, the number just before @voipjet in the Dial string is your VoipJet userid #exten = _1NXXNXX,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ   exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com NANPA   exten = _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ.   exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com WORLD   ;Do not change IAX2/9333 in the above two lines!   Step 3A (recommended): Set your codec to G.711 ulaw for optimal
 sound clarity and minimal transmission delay. In iax.conf (found in /etc/asterisk) locate the codec section and include the following only. disallow=all ; Prevent all codecs...   allow = ulaw ; ...except G.711 ulaw  Step 3B (recommended): Also in iax.conf, enable the jitter buffer. This section is usually immediately below the codecs section.jitterbuffer=yes ; Jitter buffer enabled...  dropcount=1 ; ...to drop at most 0.5% of VoIP packetsAre there any modifications needed in X-Lite softphone settings?  Do I need any extra hardware needed to implement this?  Please help me. Looking forward for your response.  ThanksRegards, ChandramouliChris Blunt [EMAIL PROTECTED] wrote: Hi Chandramouli Setting up VoipJet is quite simple really, you have done all the hard bit toget you Asterisk config this far.Firstly may I point out if you are posting your configuration to this listyou change your password information, as you have just given everyone accessto your account at voipjet.Make the changes to your iax.conf as voipjet suggest, the config they giveyou is generated for you and is not generic.Then you will need to add some
 provision in your dialplan (extensions.conf)to route your outbound calls.Something like:exten = _9.,1,SetCIDNum(123456432) ; This is your proper phone numberexten = _9.,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45,tr) ;dials the numberWhat this does:  To make a call dial 9 followed by the number and press dialon x-lite.  The first command sets your Caller ID number.  The second linestrips the 9 from the beginning of your number and hands the call to voipjetto terminate.You will need to ensure that your users have access to the context in wichyou put these entries.As voipjet are US based you will need to dial your numbers in a us format.Ie. 312 xxx  (for calling Chicago).Hope this helps you out.Chris___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update
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Re: [Asterisk-Users] Lighting up a light on an aastra phone

[Dave Cotton]

On Wed, 2006-05-24 at 15:55 -0400, Matt wrote:
 Hi,
 Does anyone know of some way to make a light on one of the keys of an
 aastra phone light up (or go off) by sending it a command from
 asterisk?

Now. can I work out which Aastra model you are talking about from the
above?

My guess is it's a 9133i, am I right?

-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] PCI-X PRI hardware

[David Waugh]

Hi Steve,

Eicon Networks make a 4 Port PRI card that works in a PCI-X slot.
See here for more information:
http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4pri.
htm

David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: 25 May 2006 01:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PCI-X PRI hardware

Boris Bakchiev wrote:

HI,

Does anyone know if there is a PCI-X 4 port PRI cards available on the
market?

If so, have anyone used it and how reliable they were?

Any help is appreciated...
  

The 3.3V Digium cards and the Sangoma cards work in PCI-X slots. 
However, I still haven't seen a PCI-Express card for telephony. Seems 
like they should be appearing soon, considering the way motherboards are

moving.

Regards,
Steve

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Re: [Asterisk-Users] playback windows recorded sound

[Akpome Akpoguma]

I downloaded recordPad and recorded a wav file and tried playback on 
asterisk got the same error as before -- WARNING [1225991360] 
Format.wav.c:132 check_header:unexpected header size 18--


when I recorded in gsm format on my laptop asterisk did playback well

I used sox to resample the recorded wav file on the asterisk machine into 
wav again and asterisk playback worked well.


The sound property of the recorded wav file is as follows

Bit Rate128kbps
Audio sample size   16 bit
Channels   1(mono)
Audio sample rate   8 kHz
Audio Format PCM

Is there any reason for this behaviour?



From: Doug Lytle [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] playback windows recorded sound
Date: Mon, 22 May 2006 14:12:25 -0400

Akpome Akpoguma wrote:

Hi guys,

I recorded a wav file on my windows xp laptop and tried to playback on 
asterisk but got the following error..unexpected header error 
18.when  I recorded sound using a sip phone on asterisk and compared 
with what I recorded on windows the sound property looked the 
same.does anyone have an idea how I can resolve this?



This should help:

http://www.voip-info.org/wiki/view/Asterisk+sound+files

Doug

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[Asterisk-Users] TDM2400P Problem

[Thawat Mohchit]

Hi All,

I have the problem with TDM2414E card (16FXO  4 FXS with echo
cancellation). I have already connected to the asterisk server and
loaded wctdm24xxp module. When I connect the phone to FXS port, It gets
the dial tone but I cannot dial to any number (I can press the phone
button but nothing happens even the busy tone). There is no activity in
the asterisk CLI. It just prints

asterisk*CLI set verbose 10
Verbosity was 5 and is now 10
   -- Starting simple switch on 'Zap/52-1'
   -- Hungup 'Zap/52-1'

And also the FXO port, It is connected with NEC line. I(Asterisk) can
dial to the party(NEC), they can hear all my voice but I cannot hear
anythings from them.

How can I fix these problem?  Are there any specific configuration for
TDM2400P card?

Thanks for your help.

Thawat

This is my configuration:
Zaptel.conf
loadzone = us
defaultzone = us
#Digium Wildcard TE110P T1/E1 Card
span=1,1,0,ccs,hdb3,crc4,yellow
bchan=1-15,17-31
dchan=16
#Wildcard TDM2400P Prototype Board
fxsks=32-47
fxoks=52-55

Zapata.conf
[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=unknown
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
echocancel=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callprogress=no
callerid=asreceived

; TE110P T1/E1 Card
group=1
signalling=pri_cpe
context=from-pstn
channel = 1-15,17-31; TDM2400 Card
group=2
signalling=fxs_ks
context=from-nec
callerid=asreceived
channel = 32-47

; TDM2400 Card
group=3
relaxdtmf=yes
signalling=fxo_ks
context=from-fxs
callerid=asreceived
channel = 52-55


Extension.conf
.
.
.
[from-fxs]
exten = _[12]X.,1,Dial(${TRUNK_NEC}/${EXTEN},30,r)
exten = _[12]X.,2,Hangup
.
.
.
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[Asterisk-Users] connecting asterisk to hylafax via t38modem: is it possible?

[Giorgio Incantalupo]

Hi,
I'm trying to use Hylafax without a modem. Is it possible to use 
t38modem to make Hylafax send and receive fax via Asterisk?
If yes, how? I'm searching on internet but still haven't found anything 
useful.


TIA

GIorgio Incantalupo
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RE: [Asterisk-Users] macro-dial

[Mimmus]

Hi,
I digged in dialparties.agi and found that apart from DND, hunt-group,
status, etc its main function is looking up real device(s) for any user from
AstDB. In fact, AMP/FreePBX keep a long list of entries in AstDB for any
device/user.

I'm interested in knowing how people on this list manage link between an
extension and the real device (SIP, Zap, etc).

Thanks





From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe
Lindheimer
Sent: Wednesday, May 24, 2006 7:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] macro-dial


It's not that simple. dialparties is fundamental to the whole
dialplan in AMP/freepbx and accomplishes a lot of the features such as hunt
groups, DND, etc. And extensions are not necessarily what you think they are
either. If you don't like it, you'd probably be better off writing your own
dialplan or alternatively, rewrite it's entire functionality outside of an
agi and then submit the mod to freepbx to streamline freepbx more.
 
p

From: Mimmus [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Wed, 24 May 2006 18:00:36 +0200
Subject: [Asterisk-Users] macro-dial

Hi,
I'm trying to edit an AMP-derived dialplan: the macro dial uses
the AGI
script dialparties.agi to find the extension to call.
I'd like to drop this script: does anyone can explain me what is its
main
job?

Thanks
-- 
Domenico Viggiani





How low will we go? Check out Yahoo! Messenger's low PC-to-Phone
call rates.
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Re: [Asterisk-Users] connecting asterisk to hylafax via t38modem: is it possible?

[Thomas Kenyon]

Giorgio Incantalupo wrote:
 Hi,
 I'm trying to use Hylafax without a modem. Is it possible to use
 t38modem to make Hylafax send and receive fax via Asterisk?
 If yes, how? I'm searching on internet but still haven't found
 anything useful.

Afaik, not yet, Steve Underwood who writes on the Dev mailing list is
doing a lot of work regarding integrating T.38, at the moment. The
nearest thing you can get is his Iaxmodem, which will set up a
/dev/device softmodem for hylafax to treat as a modem and parse the data
to asterisk though an iax channel (very clever stuff) where it can be
outputted through any other channel  in your dialplan. (or conversely
the other way for incoming faxes.)

I've used it extensively at home, and it works well. About to install it
at a site.


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RE: [Asterisk-Users] connecting asterisk to hylafax via t38modem: is itpossible?

[Mimmus]

I know only IAXmodem, a software doing what you want using IAX protocol.
Some guys on this list use it with some success.

Mimmo Viggiani


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Giorgio Incantalupo
 Sent: Thursday, May 25, 2006 10:10 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] connecting asterisk to hylafax via 
 t38modem: is itpossible?
 
 Hi,
 I'm trying to use Hylafax without a modem. Is it possible to 
 use t38modem to make Hylafax send and receive fax via Asterisk?
 If yes, how? I'm searching on internet but still haven't 
 found anything useful.
 
 TIA
 
 GIorgio Incantalupo

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Re: [Asterisk-Users] # key

[Akpome Akpoguma]

I was actually running record() application, when I pressed the # key to 
interrupt the recording, it just doesnt stop



From: Lacy Moore - Aspendora [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] # key
Date: Wed, 24 May 2006 12:40:44 -0500

I believe you want the ESP mailing list.  Of the two message I have read
that you have posted today, both require mind reading abiltities to answer
your questions.

Please include a little more information.


On 5/24/06, Akpome Akpoguma [EMAIL PROTECTED] wrote:



When I use the # key to interrupt an application it does not work.
Pls is there any idea on what could be wrong?

Rgds,

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Aspendora, Inc.




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Re: [Asterisk-Users] What and When is the next version of Asterisk?

[Pavel Jezek]

... and because sip/rtp jitterbuffer implementation still isn't in 
trunk, so will not be included in 1.4 release? :'(

PJ


BJ Weschke wrote:


Architectural freeze did happen some time ago. Features will still
continue to go in up until the end of May. app_followme will be one
such feature that is still to go in and will be in 1.4 betas and
release.


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[Asterisk-Users] Voice Mail Audio Progression

[Bob Chiodini]

Good Morning,

I've been trying to set up [EMAIL PROTECTED] and thing are going pretty 
well.  I do have a question:  When I *98 into voice mail I hear a 
message that says Asterisk mail then short pause then the word 
mailbox then a very long pause, then a request for a password.  I 
believe some other audio should occur in the long pause between 
mailbox and the request for a password.  Am I correct?  What should I 
hear and how can I troubleshoot?

From the log:

May 24 22:08:06 VERBOSE[4862] logger.c: -- Executing 
VoiceMailMain(SIP/201-33cc, default) in new stack
May 24 22:08:06 DEBUG[4862] channel.c: Scheduling timer at 160 sample intervals
May 24 22:08:06 VERBOSE[4862] logger.c: -- Playing 'vm-login' (language 'en')
May 24 22:08:08 DEBUG[4865] manager.c: Manager received command 'Command'
May 24 22:08:08 DEBUG[4865] manager.c: Manager received command 'Command'
May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 84 sample intervals
May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals
May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals
May 24 22:08:19 DEBUG[4862] channel.c: Scheduling timer at 160 sample intervals
May 24 22:08:19 VERBOSE[4862] logger.c: -- Playing 'vm-password' (language 'en')
May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 63 sample intervals
May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals
May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals
May 24 22:08:23 WARNING[4862] app_voicemail.c: Unable to read password
May 24 22:08:23 DEBUG[4862] pbx.c: Extension *98, priority 5 returned normally 
even though call was hung up
May 24 22:08:23 VERBOSE[4862] logger.c: -- Executing Macro(SIP/201-33cc, 
hangupcall) in new stack

Any hints?

Bob...


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[Asterisk-Users] Asterisk MySQL

[Abdul Lateef]

Hi all,

I am using MySQL query inside my extentions.conf. i
have more than 200 agents using the same extentions
and i can see in each request asterisk try to connect
mysql. 

My question is, Is there any way to make only one
connection for all users who is using the same
extentions.

Here is my example working extentions:

[mysqlt]
exten = _X.,1,MYSQL(Connect connid 192.168.1.65
username password database)
exten = _X.,2,MYSQL(Query r ${connid} INSERT\ INTO\
Userstabl\ set\ user=921)
exten = s,n,MYSQL(Disconnect ${connid})

Please advice me how i can make one connection for all
users?


Thank You
Abdul

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[Asterisk-Users] Re: Placing call files in /var/spool/asterisk/outgoing/ does not work

[Tony Mountifield]

In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Wed, May 24, 2006 at 03:06:54PM +0200, Maxim Vexler wrote:
  Hello everyone
  
  I'm trying to make asterisk get a call out using the .call system.
  The setup is [EMAIL PROTECTED] 2.6
  
  This is the content of the file is :
  
  Channel: Zap/g0/052MYPHONE
  MaxRetries: 2
  RetryTime: 60
  WaitTime: 30
  #
  # Assuming that your local extensions are kept in the
  #  context called [extensions]
  #
  Context: ext-local
  Extension: 210
  Priority: 1
  
  
  I'm coping (as root) from /root/call to 
  /var/spool/asterisk/outgoing/max.call
 
 you should mv the file (and in the same filesystem, so 'rename' is used)

True.

  This is what tunes up in the console :
  
  May 24 08:57:27 WARNING[10618]: pbx_spool.c:347 scan_service: Unable
  to open /var/spool/asterisk/outgoing/max.call: Permission denied,
  deleting
  May 24 08:57:27 WARNING[10618]: pbx_spool.c:389 scan_thread: Failed to
  scan service '/var/spool/asterisk/outgoing/max.call'
  
  
  What am I doing wrong ?
 
 Letting asterisk read it before it is complete.

That's not what Permission denied suggests.

I don't know AAH, but if it runs asterisk as a non-root user, then it
is necessary to chown the call file to the user under which asterisk
runs, before moving it into the outgoing directory. Or to chmod it
to 666.

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Voice Mail Audio Progression

[Avi Miller]

On 25/05/2006, at 8:14 PM, Bob Chiodini wrote:


message that says Asterisk mail then short pause then the word
mailbox then a very long pause, then a request for a password.  I


Its asking you for your mailbox number at that point, then pausing to  
allow you to enter the mailbox number. When you don't, it assumes you  
mean the mailbox associated with the extension you're dialling in from.


Hope that helps,
Avi

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[Asterisk-Users] VLAN info

[Alan Neville]

Title: VLAN info






Hello,

I'm looking for any information on setting up VLANs to seperate the telephony network from the ordinary network. I have google'd around but haven't found a lot of information in the best way to go about this. Has anyone managed to do this successfully in conjunction with asterisks? If so, could they provide an overview of what they did and how they find it? Did the performance improve after the VLANs were setup?

Many Thanks,


Alan Neville

Barlan Technologies,
Technical Support and Helpdesk,
[EMAIL PROTECTED]
+353 1 866 6111





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Re: [Asterisk-Users] What and When is the next version of Asterisk?

[BJ Weschke]

On 5/25/06, Pavel Jezek [EMAIL PROTECTED] wrote:

... and because sip/rtp jitterbuffer implementation still isn't in
trunk, so will not be included in 1.4 release? :'(


They were working on it pretty actively on Tuesday, but they were
still having issues when someone tested it on the dev conf call. It
has until the end of the month to get in.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Voice Mail Audio Progression

[Bob Chiodini]

I don't hear a request for my mailbox number. Should it say something 
like Enter mailbox number?


Bob...

Avi Miller wrote:


On 25/05/2006, at 8:14 PM, Bob Chiodini wrote:


message that says Asterisk mail then short pause then the word
mailbox then a very long pause, then a request for a password. I


Its asking you for your mailbox number at that point, then pausing to 
allow you to enter the mailbox number. When you don't, it assumes you 
mean the mailbox associated with the extension you're dialling in from.


Hope that helps,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore Street T: +61 (0) 3 9235 5400
Fitzroy, VIC F: +61 (0) 3 9235 5444
3065 W: http://www.squiz.net

.   Open Source - Own It - Squiz.net .. /



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Re: [Asterisk-Users] Voice Mail Audio Progression

[Avi Miller]

On 25/05/2006, at 8:57 PM, Bob Chiodini wrote:

I don't hear a request for my mailbox number. Should it say  
something like Enter mailbox number?


I believe the prompt just goes Mailbox? -- its not great. But,  
there's no other prompts being played in your output.


--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
  2/340 Gore StreetT: +61 (0) 3 9235 5400
  Fitzroy, VIC F: +61 (0) 3 9235 5444
  3065 W: http://www.squiz.net

.   Open Source - Own It - Squiz.net .. /




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Re: [Asterisk-Users] What and When is the next version of Asterisk?

[Pavel Jezek]

I think, that sip/rtp jitterbuffer is one of the most wanted feature, 
but because still not included in trunk too few peoples improving it...
what to try include this soon to trunk, and only if problems will be not 
solved before 1.4 release candidate, remove out of asterisk 1.4 ...
also good candidate to 1.4 is new codec negotiation algorithm, seems be 
actively maintained/finalized

http://bugs.digium.com/view.php?id=4825
PJ


BJ Weschke wrote:

On 5/25/06, Pavel Jezek [EMAIL PROTECTED] wrote:

... and because sip/rtp jitterbuffer implementation still isn't in
trunk, so will not be included in 1.4 release? :'(


They were working on it pretty actively on Tuesday, but they were
still having issues when someone tested it on the dev conf call. It
has until the end of the month to get in.


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Re: [Asterisk-Users] playback windows recorded sound

[Time Bandit]

I downloaded recordPad and recorded a wav file and tried playback on
asterisk got the same error as before -- WARNING [1225991360]
Format.wav.c:132 check_header:unexpected header size 18--

when I recorded in gsm format on my laptop asterisk did playback well

I used sox to resample the recorded wav file on the asterisk machine into
wav again and asterisk playback worked well.

The sound property of the recorded wav file is as follows

Bit Rate128kbps
Audio sample size   16 bit
Channels   1(mono)
Audio sample rate   8 kHz
Audio Format PCM

Is there any reason for this behaviour?

See this page :
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

hth
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Re: [Asterisk-Users] What and When is the next version of Asterisk?

[BJ Weschke]

On 5/25/06, Pavel Jezek [EMAIL PROTECTED] wrote:

I think, that sip/rtp jitterbuffer is one of the most wanted feature,
but because still not included in trunk too few peoples improving it...
what to try include this soon to trunk, and only if problems will be not
solved before 1.4 release candidate, remove out of asterisk 1.4 ...
also good candidate to 1.4 is new codec negotiation algorithm, seems be
actively maintained/finalized
http://bugs.digium.com/view.php?id=4825
PJ


PJ,

I understand what you're saying, but playing devil's advocate, if
this goes into /trunk it becomes part of the 1.4 release. The
community is then tasked with supporting this feature, whether or not
it is ready for prime time. If it's not ready, people then complain
that features are in the release that don't work and weren't
tested. It's kind of a no-win situation.

The best thing that can be done at this point to work towards trying
to get this feature in is to have people test it in /trunk in their
environments and report results back. If they are developers and can
contribute to code improvements around it, that would be very much
welcomed as well.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] DHCP configuration for Cisco 7960?

[Neil Cherry]

Julian Dunn wrote:

(Apologies to those Toronto Asterisk Users' Group folks who have seen
this message... I figured I'd have more success with a wider audience)


I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on
FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on
the wire. I wonder if anyone has done this before and therefore can
validate whether or not the traffic I am seeing is normal.

I have dhcpd.conf set up like this

---8--- cut here ---8---

option cisco-etherboot-server code 150 = ip-address; .
.
.
host c7960 {
hardware ethernet 00:16:46:9B:6D:62;
fixed-address 192.168.5.14;
option host-name c7960.acf.aquezada.com;
option cisco-etherboot-server 192.168.5.7; }



host c7960 {
hardware ethernet  00:0d:11:22:33:44;
fixed-address  192.168.2.196;
option domain-name uucp ;
option tftp-server-name192.168.2.1;
always-reply-rfc1048   true ;
}

Here's mine, works under Linux.

--
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://www.linuxha.com/ Main site
http://linuxha.blogspot.com/My HA Blog
http://home.comcast.net/~ncherry/   Backup site
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Re: [Asterisk-Users] soekris hadware

[Woodoo People .pGa!]

 i'm brand new and i would like to ask about soekris hardware. I read
 along the web but i have some doubts that i think can be solved here.
 My question are the following:
personally i have no experience, but i think you have to forget g.729,
and also handling more than 2-3 paralell calls.
 
 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a
 bigger box is needed? Any suggestions about where to pick up another
 box?
 
 2) does the Digium TDM100P (already discontinued) fits fine in a soekris 
 box?
 
 3) running asterisk in a soekris 4801 SBC, what is the perfomance
 related to sip connections, analogue call quality and both mixed at
 the same time?
 
 4) what is the actual state of fax support into asterisk / [EMAIL PROTECTED] ?
it's working for me in and out.

 5) It's posible to create personalized dialplans that enables a hidden
 or passcode/password protected menu for remote administration or
 remote use of the pbx?
yes
-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Database Integration

[Thomas Kenyon]

Bruno de Assumpção Loureiro wrote:
 On 5/23/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
 Bruno de Assumpção Loureiro wrote:
  Hi all,
 
  How to integrate with Oracle database. I think it's possible with AGI,
  it isn't?
 
  Regards,
 
 You will probably need to go down the odbc route for oracle.

 Read the details on cdr_odbc.conf, extconfig.conf, res_config_odbc.conf
 and res_odbc.conf on voip-info.org.

 I think these confs  are only to store the configuration files into an
 OBDC database. or Is able to integration an Oracle database (not the
 asterisk configuration files)  to asterisk box with these confs?

 I need to read, write and del  from tables at an Oracle Server, which
 is working in another workstation.


For actions in a dialplan, look at DBGet, DBPut and DBDel or app_dbodbc
at http://www.voip-info.org .

For AGI scripts, you can access them, the same way you normally do
(ODBC/DBI) or using the AGI database commands.

 Asterisk  AGI script  OBDC  Oracle Server

 Is it right? There is another way? Somenone has it working?

Yeah lots :-)

I'm surprised no-one else has jumped on this one yet, I'm certainly no
expert.

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[Asterisk-Users] Re: Voice Mail Audio Progression

[Steven]

It would be nice if there was a .conf file for the voicemail prompts.
I you rerecorded Mailbox? it may break other announcement that might use that 
file.
But, if there were a .conf file with the prompts listed were you could change 
them, like mailboxprompt=voicemail/mailbox, you could 
change these at will.
Also, if the prompts were categorized in some way, you could have one 
extensions for English prompts, one extension for German, etc.

-- 
-- 
Steven

http://www.glimasoutheast.org



Avi Miller [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]

 On 25/05/2006, at 8:57 PM, Bob Chiodini wrote:

 I don't hear a request for my mailbox number. Should it say  something like 
 Enter mailbox number?

 I believe the prompt just goes Mailbox? -- its not great. But,  there's no 
 other prompts being played in your output.

 --
 National Manager - Special Projects

  Sydney / Melbourne / Canberra / Hobart / London /
   2/340 Gore StreetT: +61 (0) 3 9235 5400
   Fitzroy, VIC F: +61 (0) 3 9235 5444
   3065 W: http://www.squiz.net

 .   Open Source - Own It - Squiz.net .. /




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Re: [Asterisk-Users] Voice Mail Audio Progression

[Bob Chiodini]

Avi,

Got it!  Thanks. The minimalistic approach :-)

Bob...

On Thu, 2006-05-25 at 21:07 +1000, Avi Miller wrote:
 On 25/05/2006, at 8:57 PM, Bob Chiodini wrote:
 
  I don't hear a request for my mailbox number. Should it say  
  something like Enter mailbox number?
 
 I believe the prompt just goes Mailbox? -- its not great. But,  
 there's no other prompts being played in your output.
 
 --
 National Manager - Special Projects
 
  Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore StreetT: +61 (0) 3 9235 5400
Fitzroy, VIC F: +61 (0) 3 9235 5444
3065 W: http://www.squiz.net
 
 .   Open Source - Own It - Squiz.net .. /
 
 
 
 
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Re: [Asterisk-Users] Video SIP Softset

[Thomas Kenyon]

Curt Shaffer wrote:

 Sorry if this shows twice but it appears my first message was
 quarantined because of my digital signature.

  

 All,

  

 I have been tasked with setting up video conferencing utilizing
 asterisk. One of the requirements is a softset that has video
 capabilities. Eyebeam looks promising but I was just wondering if
 anyone out there knows of any freeware with comparable features of
 Eyebeam that they have used successfully with Asterisk.

  

 Thanks

  

 Curt

  

If you're using windows XP, then Netmeeting will already be installed
(Start - Run - Conf) and can be integrated into asterisk (Using H.323
rather than SIP).

Have you tried all the usual places (sourceforge, freshmeat etc.?)




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Re: [Asterisk-Users] Permit/Deny function question

[Rich Adamson]

Kevin Kiely wrote:

I am trying to limit IAX connectivity to a server with the permit/deny
combination.  In this example to allow ip 123.123.123.123 but it's not
working.  If I remove the mask on the deny parameter it allows all
hosts.  With the deny statement like below it blocks all connections
even using a mask or no mask with the permit IP.  What am I missing?



allow=all
context=from-external
secret=sample
type=user
deny=0.0.0.0/0.0.0.0
permit=123.123.123.123


The above should be written something like:
deny=0.0.0.0/0.0.0.0
permit=123.123.123.123/255.255.255.0

R.



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Re: [Asterisk-Users] VLAN info

[Florian Overkamp]

Hi,

Citeren Alan Neville [EMAIL PROTECTED]:

 I'm looking for any information on setting up VLANs to seperate the telephony
 network from the ordinary network. I have google'd around but haven't found a
 lot of information in the best way to go about this. Has anyone managed to do
 this successfully in conjunction with asterisks? If so, could they provide an
 overview of what they did and how they find it? Did the performance improve
 after the VLANs were setup?

Using VLAN's (and more importantly, VLAN priority settings) will most definitely
be able to improve VoIP quality. In Linux, a VLAN will be another logical
ethernet interface, and thus, to the configuration of Asterisk it makes no
difference. Take a look at:

http://www.linuxjournal.com/article/7268

-- 
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
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Re: [Asterisk-Users] Permit/Deny function question

[Patrick]

On Thu, 2006-05-25 at 00:44 -0400, Kevin Kiely wrote:
 I am trying to limit IAX connectivity to a server with the permit/deny
 combination.  In this example to allow ip 123.123.123.123 but it's not
 working.  If I remove the mask on the deny parameter it allows all
 hosts.  With the deny statement like below it blocks all connections
 even using a mask or no mask with the permit IP.  What am I missing?
 
 allow=all
 context=from-external
 secret=sample
 type=user
 deny=0.0.0.0/0.0.0.0
 permit=123.123.123.123

You need to add a netmask to the permit line too. Try
permit=123.123.123.123/255.255.255.0 or whatever netmask you need.

Regards,
Patrick

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RE: [Asterisk-Users] VLAN info

[Mimmus]

Title: VLAN info



We are currently using VLANs for all our 
networks.
Setuping a VoIP VLAN was simply a matter of configuring 
some switches. Defining higher 802.1p priority for switch ports on this VLAN was 
the following logical step.
We don't 
use VLAN tag on the phones directly.

DV

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Alan 
  NevilleSent: Thursday, May 25, 2006 12:44 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] VLAN 
  info
  
  Hello,I'm looking for any information on setting up 
  VLANs to seperate the telephony network from the ordinary network. I have 
  google'd around but haven't found a lot of information in the best way to go 
  about this. Has anyone managed to do this successfully in conjunction with 
  asterisks? If so, could they provide an overview of what they did and how they 
  find it? Did the performance improve after the VLANs were setup?Many 
  Thanks,Alan NevilleBarlan Technologies,Technical 
  Support and Helpdesk,[EMAIL PROTECTED]+353 1 866 
6111
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Re: [Asterisk-Users] Lighting up a light on an aastra phone

[Matt]

Yes sorry.. it is a 9133i.. it seems I cannot.. but if you have a
suggest for how to light a light please let me know.

On 5/25/06, Dave Cotton [EMAIL PROTECTED] wrote:

On Wed, 2006-05-24 at 15:55 -0400, Matt wrote:
 Hi,
 Does anyone know of some way to make a light on one of the keys of an
 aastra phone light up (or go off) by sending it a command from
 asterisk?

Now. can I work out which Aastra model you are talking about from the
above?

My guess is it's a 9133i, am I right?

--
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] Anyone got a used T1 card I can have?

[Mark Phillips]

Hi folks,

Anyone got a gently used working T1 card I can have? 

Can pay by CC, check, cheque or Paypal.

Thanks

Mark

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Re: [asterisk-biz] RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP

[Erick Perez]

 an
   AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit).
   I purchased them second-handed with no manuals (thank god for the
   internet!!) but i guess the pdf manual I have does not have the
   section of factory-reset.
  
   Also, any sucess stories with:
   AudioCodes MP124-C/FSX/AC/SIP
  ---Asterisk---internet---Vonage
   setups?
  
   Thanks,
  
  
   --
  
   ---
   Erick Perez
   Linux User 376588
   http://counter.li.org/  (Get counted!!!) Panama, Republic
 of Panama
   ___
 
 
 
 

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--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] VLAN info

[mohamed kerbachi]

Hi Mimmus,

Can you give us some exemples about Defining higher
802.1p priority for switch  ?,
any URLs with tutorial will be welcome.

Thx a lot.


--- Mimmus [EMAIL PROTECTED] a écrit :

 We are currently using VLANs for all our networks.
 Setuping a VoIP VLAN was simply a matter of
 configuring some switches.
 Defining higher 802.1p priority for switch ports on
 this VLAN was the
 following logical step.
 We don't use VLAN tag on the phones directly.
  
 DV
 
 
   _  
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Alan Neville
 Sent: Thursday, May 25, 2006 12:44 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] VLAN info
 
 
 
 Hello,
 
 I'm looking for any information on setting up VLANs
 to seperate the
 telephony network from the ordinary network. I have
 google'd around but
 haven't found a lot of information in the best way
 to go about this. Has
 anyone managed to do this successfully in
 conjunction with asterisks? If so,
 could they provide an overview of what they did and
 how they find it? Did
 the performance improve after the VLANs were setup?
 
 Many Thanks,
 
 
 Alan Neville
 
 Barlan Technologies,
 Technical Support and Helpdesk,
 [EMAIL PROTECTED]
 +353 1 866 6111
 
 
 
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Re: [Asterisk-Users] soekris hadware

[Jonathan Gonzalez]

Thanks a lot for all your post... any other will be welcomed.
Kind regards,

Jonathan GF


On 5/25/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote:

 i'm brand new and i would like to ask about soekris hardware. I read
 along the web but i have some doubts that i think can be solved here.
 My question are the following:
personally i have no experience, but i think you have to forget g.729,
and also handling more than 2-3 paralell calls.

 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a
 bigger box is needed? Any suggestions about where to pick up another
 box?

 2) does the Digium TDM100P (already discontinued) fits fine in a soekris
 box?

 3) running asterisk in a soekris 4801 SBC, what is the perfomance
 related to sip connections, analogue call quality and both mixed at
 the same time?

 4) what is the actual state of fax support into asterisk / [EMAIL PROTECTED] ?
it's working for me in and out.

 5) It's posible to create personalized dialplans that enables a hidden
 or passcode/password protected menu for remote administration or
 remote use of the pbx?
yes
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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--
si secretum tibi sit, tege illud, vel revela
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Re: [Asterisk-Users] TE406P - MFC/R2

[Fernando Lujan]

Steve Underwood wrote:


Hi Steve, nice to hear from you. I spoke with you in the irc some time ago.




I'm trying to configure a TE406P with MFC/R2.

here goes my zaptel.conf:

span=1,0,0,ccs,hdb3,crc4
cas=1-15:1101
dchan=16
cas=17-31:1101

span=2,0,0,ccs,hdb3,crc4
cas=32-46:1101
dchan=47
cas=48-62:1101


Changed to:

span=1,1,0,cas,hdb3
span=2,0,0,cas,hdb3
#
cas=1-15:1101
cas=17-31:1101
#
cas=32-46:1101
cas=48-62:1101


That config is completely wrong. Try following the config in the 
MFC/R2 documentation. And please use the documentation at 
soft-switch.org. I keep getting problems reported by people using some 
random junk documentation they found somewhere else.




The first strange behavior is that the:

zap show status

shows this:

Description  Alarms IRQ
bpviol CRC4
T4XXP (PCI) Card 0 Span 1OK 0  
0  0
T4XXP (PCI) Card 0 Span 2OK 0  
0  0
T4XXP (PCI) Card 0 Span 3OK 0  
0  0
T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0  
0  0


How is it possible? Since i just configure 2 spans!


If you configured chan_zap for 2 spans, this is also wrong. You should 
be configuring chan_unicall for the spans. Again, look at the examples 
in he documentation.

My zapata.conf is empty. I used unicall.conf instead.

Here it goes:

[channels]

context=incoming
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
amaflags=billing
musiconhold=default
loglevel=255
protocolclass=mfcr2
protocolvariant=br,20,12
group = 1
context=incoming
channel = 1-15
channel = 17-31
channel = 32-46
channel = 48-62


Applying the pach in the Makefile just failed. So I do this:

Change:

chan_unicall.o: chan_unicall.c
  $(CC) -c $(CFLAGS) -o chan_unicall.o chan_unicall.c

To:

chan_unicall.so: chan_unicall.o
  $(CC) $(SOLINK) -o $@ $ -lunicall -lxml2 -lsupertone -lspandsp -ltiff 
$(ZAPLIB)

Change:

CHANNEL_LIBS=chan_sip.so chan_agent.so chan_mgcp.so chan_iax2.so chan_local.so 
chan_skinny.so chan_features.so

To:

CHANNEL_LIBS=chan_sip.so chan_agent.so chan_mgcp.so chan_iax2.so chan_local.so 
chan_skinny.so chan_features.so chan_unicall.so

And finally:

cp channels/chan_unicall.so /usr/lib/asterisk/modules/

Thnaks in advance.

Fernando Lujan




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[Asterisk-Users] IVR transcoding g729 license

[Tamas]

Hello,

- a call comes in with protocol SIP, codec g729
- connects to the IVR [playing prompts, collecting digits]
- dialing forward to protocol IAX2, codec g729

I guess, while the call is in IVR, I need one g729 license for the call.
If the call leaves the IVR, thus dials to another box with the same g729
codec, will it need 2 licenses? While there should be no transcoding
further, will it free up the codec which was used during IVR session?

I would like to avoid useless transcoding [g729-slin-g729] in case the
call goes through an IVR.

How is such case handled in asterisk? [1.2]

Thanks in advance,
Tamas
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[Asterisk-Users] Error on Polycom 501 601.

[Ken D'Ambrosio]

Hi, all.  Every now and then, some of my users get Error on their
phones.  A reboot fixes it, but it's quite annoying/inconvenient.  I'm
running Asterisk 1.2.4, and have the following firmware, etc.:

Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041

Any ideas as to why this might be happening?

Thanks!

-Ken

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Re: [Asterisk-Users] Generate two calls from Asterisk and bridge them

[Nicolás Gudiño]

You can also try this patch:

http://bugs.digium.com/view.php?id=5841

On 5/24/06, Arjan Kroon [EMAIL PROTECTED] wrote:

I recommended simple Meetme conference bridge

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


Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Álvaro Palma
Sent: woensdag 24 mei 2006 16:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Generate two calls from Asterisk and bridge them

Is there a way in Asterisk (I guess there's, it's only I can't figure
out how :-)) to:

1.- Generate a call to channel 1 (example, to PSTN vía an E1 card, using
Zap/g1)

2.- Generate a call to channel 2 (example, an internal SIP extension).

3.- Once both channel have answered, connect the call between them.

This way, I can, for example, play audios in both channels before they
are connected between each other.

So my question is: Does anybody figures out a way to do this? If I use
Manager/Originate, the call necesarily needs a channel to be picked up
(the originating channel) before the call can be placed. What I'd like
to do is:

Asterisk - Channel 1 and do something in channel 1
Asterisk - Channel 2 and do something in channel 2

Bridge both channels: Channel 1  Channel 2

Is maybe Local the solution?

Thanks a lot for your help.

--
Atly.
Alvaro Palma
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--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Video SIP Softset

[Matt Riddell (IT)]

Curt Shaffer wrote:
 I have been tasked with setting up video conferencing utilizing asterisk.
 One of the requirements is a softset that has video capabilities. Eyebeam
 looks promising but I was just wondering if anyone out there knows of any
 freeware with comparable features of Eyebeam that they have used
 successfully with Asterisk.

Try the Asterisk-Video mailing list.  Details are available here:

http://lists.digium.com/mailman/listinfo/asterisk-video

Apparently there is Wengo and Kapanga which may or may not suit your needs.

-- 
Cheers,

Matt Riddell
___

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http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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[Asterisk-Users] CallerID from cell phone not being rewritten

[Kyle Sexton]

I'm having a problem with my dialplan. I'm trying to rewrite the CallerID Name variable so that when a call comes in, it shows what queue the call is going to:exten = 1234,n,Set(CALLERID(name)=Queue1)This works fine for most calls, but when I call from my cell phone my name appears and it doesn't get rewritten for some reason. Has anyone else experienced this behavior?
Thanks,Kyle
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Re: [Asterisk-Users] macro-dial

[picciuX]

ehm... in the dialplan!Well, in some db tables, from which is generated a dialplan fragment by means of PHP scripts.2006/5/25, Mimmus [EMAIL PROTECTED]
:Hi,I digged in dialparties.agi and found that apart from DND, hunt-group,
status, etc its main function is looking up real device(s) for any user fromAstDB. In fact, AMP/FreePBX keep a long list of entries in AstDB for anydevice/user.I'm interested in knowing how people on this list manage link between an
extension and the real device (SIP, Zap, etc).ThanksFrom: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of PhilippeLindheimerSent: Wednesday, May 24, 2006 7:42 PMTo: 
asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] macro-dialIt's not that simple. dialparties is fundamental to the wholedialplan in AMP/freepbx and accomplishes a lot of the features such as hunt
groups, DND, etc. And extensions are not necessarily what you think they areeither. If you don't like it, you'd probably be better off writing your owndialplan or alternatively, rewrite it's entire functionality outside of an
agi and then submit the mod to freepbx to streamline freepbx more.pFrom: Mimmus [EMAIL PROTECTED]To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.comDate: Wed, 24 May 2006 18:00:36 +0200Subject: [Asterisk-Users] macro-dialHi,
I'm trying to edit an AMP-derived dialplan: the macro dial usesthe AGIscript dialparties.agi to find the extension to call.I'd like to drop this script: does anyone can explain me what is its
mainjob?Thanks--Domenico ViggianiHow low will we go? Check out Yahoo! Messenger's low PC-to-Phone
call rates.http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*
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Re: [Asterisk-Users] Error on Polycom 501 601.

[Johann]

Are you using an idle webpage?  If for some reason the phone can't reach the 
page it will display an error and rebooting is about the only way to fix it.


--johann

Ken D'Ambrosio wrote:

Hi, all.  Every now and then, some of my users get Error on their
phones.  A reboot fixes it, but it's quite annoying/inconvenient.  I'm
running Asterisk 1.2.4, and have the following firmware, etc.:

Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041

Any ideas as to why this might be happening?

Thanks!

-Ken

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Re: [Asterisk-Users] IVR transcoding g729 license

[Kristian Kielhofner]

Tamas wrote:

Hello,

- a call comes in with protocol SIP, codec g729
- connects to the IVR [playing prompts, collecting digits]
- dialing forward to protocol IAX2, codec g729

I guess, while the call is in IVR, I need one g729 license for the call.
If the call leaves the IVR, thus dials to another box with the same g729
codec, will it need 2 licenses? While there should be no transcoding
further, will it free up the codec which was used during IVR session?

I would like to avoid useless transcoding [g729-slin-g729] in case the
call goes through an IVR.

How is such case handled in asterisk? [1.2]

Thanks in advance,
Tamas


Tamas,

	Just record your IVR prompts in g729 (or convert them to g729 from some 
other format)!


--
Kristian Kielhofner
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[Asterisk-Users] Re: Implementing Paging on the Linksys SPA9XX phones (working)

[JR Richardson]

I came up with this a few days ago, mostly used the wiki examples,
didn't have time to post on the wiki yet, maybe one of you guys with a
few minutes can throw it up there, really, I forgot my logon.

http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom

The agi script didn't work for me, wouldn't call the active hint
extensions, even though they were there, no time to debug it so I
setup direct paging in the dialplan by setting the SIPAddHeader cmd
for each phone type.

;pages a Polycom 601 (must setup polycom sip.cfg file according to wiki)
exten = *1001,1,SIPAddHeader(Alert-Info: Ring Answer)
exten = *1001,2,Dial(SIP/1001)
exten = *1001,3,Hangup


;pages a Linksys SPA942
exten = *1003,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = *1003,2,Dial(SIP/1003)
exten = *1003,3,Hangup


;pages a Snom 320
exten = *1005,1,SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)
exten = *1005,2,Dial(SIP/1005)
exten = *1005,3,Hangup

;this group page only works for the linksys and polycom at the same time, due
;to the snom also looking for Call-Info header but a little different
than the linksys
;so can't set both.  Also Page cmd only fully functional in 1.2.7.1,
earlier versions
;the phones would not hangup after the caller hangs up, the phones kept ringing
;and other wierd issues, jsut wouldn't work correctly but 1.2.7.1
worked right away
exten = *22,1,SIPAddHeader(Alert-Info: Ring Answer)
exten = *22,1,SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)
exten = *22,2,SIPAddHeader(Call-Info:\;answer-after=0)
exten = *22,3,Page(SIP/1003SIP/1001|)

The customer implementation is with a Polycom 601 with 2 operator
panels for the receptioninst phone and Linksys 942's for the office
staff.

The receptionist can push the buddy watch position on the Polycom
console to call the linksys extensions directly or *exten to page
them, or *22 to page all extensions, works quite well.  But of course
this is a hack, true paging needs to be multicast.

Maybe putting a a page variable in the sip.conf phone profile could
set the sip header properly when the page is called in the dial plan,
like:

[1001] ;Polycom 601
page=SIPAddHeader(Alert-Info: Ring Answer)

[1003] ;Linksys SPA942
page=SIPAddHeader(Call-Info:\;answer-after=0)

[1005] ;Snom 320
page=SIPAddHeader(Call-Info: sip:10.10.11.255\; answer-after=0)

This way, you would not have to set the sip header in the dialplan,
just call the page cmd and the sip header is set per the phone called
in the group.  This would allow for different phones to be called at
the same time.

Hope this helps.

JR

--
JR Richardson
Engineering for the Masses
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Re: [Asterisk-Users] Problems with Park and MOH

[Bob McDowell]

Can I ask how you set the parking lots to a different moh? I don't see a setting for it in features.conf...
On 5/23/06, Doug Lytle [EMAIL PROTECTED] wrote:
Michael Knill wrote: I have not installed mpg123 so it must be native.
I've had issues with Native MOH over IAX2 while parking.Very distortedMOH and high pitch squealing.But, Native MOH from a PSTN or just plainputting a call on hold via the SIP phones is fine.I moved the Parking
Lots to MPG123 and the rest to Native.Doug-- Ben Franklin quote: Those who would give up Essential Liberty topurchase a little Temporary Safety, deserve neither Liberty nor Safety.
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RE: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

Domenico,as I mentioned: "...and extensions are not necessarily what you think they areeither." AMP/Freepbx 'virtualizes' extensions. The basic concept is that there are users and then there are devices. A user can have multiple devices. The default shipping mode provides the 'extensions' tab which ends up creating a user with the sam extension number as the device that you assign them. However, if you flip to 'deviceanduser' mode (see /etc/amportal.conf - AMPEXTENSIONS=) you will see that you now can control users separate from devices and you can assign multiple devices to a single user or you can make a device adhoc allowing any user to login to the device and it becomes their phone until they logout.So as I mentioned, it isn't that simple, it is the reason for all the various callerid macros, dialparties.agi, etc. that is there. If you want more detail, in addition to digging in as you have, you
 may want to move over to the freepbx.org site and/or the IRC. If you want to get rid of dialparties, maybe you can get the entire functionality into a dialplan format (and probably improve performance) and then submit it back. But as you've probably seen, dialparties itself is inegrally interwoven with macro-dial and the various other interdependencies throughout the dial plan, astdb, etc.p  From: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 10:21:46 +0200Subject: RE: [Asterisk-Users] macro-dialHi,I digged in dialparties.agi and found that apart from DND, hunt-group,status, etc its main function is looking up real device(s) for any user fromAstDB. In fact, AMP/FreePBX keep a
 long list of entries in AstDB for anydevice/user.I'm interested in knowing how people on this list manage link between anextension and the real device (SIP, Zap, etc).ThanksFrom: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of PhilippeLindheimerSent: Wednesday, May 24, 2006 7:42 PMTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] macro-dialIt's not that simple. dialparties is fundamental to the wholedialplan in AMP/freepbx and accomplishes a lot of the features such as huntgroups, DND, etc. And extensions are not necessarily what you think they areeither. If you don't like it, you'd probably be better off writing your owndialplan or alternatively, rewrite it's entire functionality outside of anagi and then submit the mod to freepbx to streamline freepbx
 more.pFrom: "Mimmus" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Date: Wed, 24 May 2006 18:00:36 +0200Subject: [Asterisk-Users] macro-dialHi,I'm trying to edit an AMP-derived dialplan: the macro "dial" usesthe AGIscript "dialparties.agi" to find the extension to call.I'd like to drop this script: does anyone can explain me what is itsmainjob?Thanks-- Domenico Viggiani
		Blab-away for as little as 1¢/min. Make  PC-to-Phone Calls using Yahoo! Messenger with Voice.___
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RE: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501phone

[broadbandvoice]

Could not find your post for 4 months ago.

-- Original message -- From: "Anton Krall" [EMAIL PROTECTED]  Yes, check a post that I made about 4 months ago, I posted the cofig for  setting the speaker, handset and ring volumes ..   |-Original Message-  |From: [EMAIL PROTECTED]  |[mailto:[EMAIL PROTECTED] On Behalf Of  |Jerry Jones  |Sent: Thursday, May 04, 2006 3:15 PM  |To: Asterisk Users Mailing List - Non-Commercial Discussion  |Subject: Re: [Asterisk-Users] Volume configuration on Polycom  |Soundpoint 501phone  |  |Edit your config files to enable persistance  |  |Will remain across multiple calls, but not reboots  |  |  |On May 4, 2006, at 2:51 PM, Jim Freeze wrote:  | 
  |
 We are using the polycom 501 phones, and are having some challenges  | with the volume setting. When a phone call comes in, the  |user ups the  | volume for the handset, but they have to repeat that for every call.  |  | Currently, the volume level seems to reset itself at about 60%.  | Is there a way for the user to change their default volume level?  |  | Thanks  |  | --  | Jim Freeze  | ___  | --Bandwidth and Colocation provided by Easynews.com --  |  | Asterisk-Users mailing list  | To UNSUBSCRIBE or update options visit:  | http://lists.digium.com/mailman/listinfo/asterisk-users  |  |___  |--Bandwidth and Colocation provided by Easynews.com --  |  
 |Aster
isk-Users mailing list  |To UNSUBSCRIBE or update options visit:  | http://lists.digium.com/mailman/listinfo/asterisk-users  |  |   ___  --Bandwidth and Colocation provided by Easynews.com --   Asterisk-Users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

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[Asterisk-Users] Glueing apps and phones together

[Julian Lyndon-Smith]

We have about 50 agents making outbound and receiving inbound calls via 
queues. All Agents have Cisco 7940 phones, using SIP (the app and the 
phones are on separate LANS).


I was wanting find a better way to glue our application (which runs on 
MS Terminal Server 2003) to our * system. At the moment we are using the 
Asterisk Manager and astmanproxy.


What I was thinking was using the new res_jabber module, a wildfire 
server and the ipworks jabber activex. Thus, * and our app can send a 
messages to each other via the wildfire server.


However, there is a little thought running through my mind about 
embedding a SIP activeX control into our app, and having * talk directly 
to the app. When our app receives or makes a call, it is automatically 
diverted / referred / transferred to the cisco phone.


I presume there are loads of people who have already achieved this type 
of glue, but was wondering if they would care to share ;)


Just looking for pointers, guidance or advice on the two methods.

TIA

Julian.
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Re: [Asterisk-Users] Is NuFone Really Dead?

[Carlos Chavez]

On Wed, 2006-05-24 at 14:00 -0700, Andy Jefferson wrote:
 Went to their site today. Site claims they are still in biz. What is
 the story? What really happened to Nufone anyway?

They say they are but I have 2 800 DIDs with them that are still
offline, they stopped working more than a month ago and all attempts to
contact support have been unanswered. I guess it is up to luck, some
users have service others do not.  Do you really want to work with a
company like that?

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


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Re: [Asterisk-Users] Is NuFone Really Dead?

[Andrew Kohlsmith]

On Thursday 25 May 2006 10:57, Carlos Chavez wrote:
  They say they are but I have 2 800 DIDs with them that are still
 offline, they stopped working more than a month ago and all attempts to
 contact support have been unanswered. I guess it is up to luck, some
 users have service others do not.  Do you really want to work with a
 company like that?

Have you gone to the members.nufone.net site and done what was asked there?  I 
don't have any DIDs myself but others have said the received notice to do 
that.

-A.
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RE: [Asterisk-Users] Is NuFone Really Dead? Same as voipjet

[Kerry Garrison]

This is the same with VoipJet, some people have good luck but my lines have
been down for 3 months and all attempts at contacting them have gone
unanswered. Hard to believe people still rave about their service. 

Here is a hint folks, if the company does not post a customer service PHONE
NUMBER don't use them. Secondly, if they do have a phone number but nobody
ever answers it, don't use them.

Just because their email address is [EMAIL PROTECTED] doesn’t mean its
fast, or is even answered. It should be /dev/[EMAIL PROTECTED]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Carlos Chavez
 Sent: Thursday, May 25, 2006 7:58 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Is NuFone Really Dead?
 
 On Wed, 2006-05-24 at 14:00 -0700, Andy Jefferson wrote:
  Went to their site today. Site claims they are still in 
 biz. What is 
  the story? What really happened to Nufone anyway?
 
   They say they are but I have 2 800 DIDs with them that 
 are still offline, they stopped working more than a month ago 
 and all attempts to contact support have been unanswered. I 
 guess it is up to luck, some users have service others do 
 not.  Do you really want to work with a company like that?
 
 --
 Carlos Chavez Prats
 Director de Tecnología
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Tel: +52-55-91169161 Ext 2001
 


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Re: [Asterisk-Users] Error on Polycom 501 601.

[Noah Miller]

Hi Ken -


 Bootrom: 2.6.2.0032
 BootBlock: 2.5.0(11500_030)
 SIP application: 1.6.2.0041


In any case, I'd suggest updating to a later firmware version.  SIP
firmware 1.6.6 has been officially released.  If you are unable to get
it, just send me a personal email (offlist).

- Noah


On 5/25/06, Johann [EMAIL PROTECTED] wrote:

Are you using an idle webpage?  If for some reason the phone can't reach the
page it will display an error and rebooting is about the only way to fix it.

--johann

Ken D'Ambrosio wrote:
 Hi, all.  Every now and then, some of my users get Error on their
 phones.  A reboot fixes it, but it's quite annoying/inconvenient.  I'm
 running Asterisk 1.2.4, and have the following firmware, etc.:

 Bootrom: 2.6.2.0032
 BootBlock: 2.5.0(11500_030)
 SIP application: 1.6.2.0041

 Any ideas as to why this might be happening?

 Thanks!

 -Ken
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Re: [Asterisk-Users] Is NuFone Really Dead?

[Carlos Chavez]

On Thu, 2006-05-25 at 11:06 -0400, Andrew Kohlsmith wrote:

 
 Have you gone to the members.nufone.net site and done what was asked there?  
 I 
 don't have any DIDs myself but others have said the received notice to do 
 that.
 
DOne that several times and I keep getting the same message.  Support
never answers the phone or email so how can I tell them it is not
working?

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


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RE: [Asterisk-Users] VLAN info

[Mimmus]

I prefer to set 802.1p priority on the switch ports instead of tagging
packets by phones capabilities.
I use only HP Procurve devices (and I'm very happy...), command is simple:

 interface 1
  qos priority 6

This priority is a VLAN tag automatically attached to packets and preserved
on uplinks.

DV


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 mohamed kerbachi
 Sent: Thursday, May 25, 2006 3:51 PM
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] VLAN info
 
 Hi Mimmus,
 
 Can you give us some exemples about Defining higher 802.1p 
 priority for switch  ?, any URLs with tutorial will be welcome.
 
 Thx a lot.
 
 
 --- Mimmus [EMAIL PROTECTED] a écrit :
 
  We are currently using VLANs for all our networks.
  Setuping a VoIP VLAN was simply a matter of configuring 
 some switches.
  Defining higher 802.1p priority for switch ports on this 
 VLAN was the 
  following logical step.
  We don't use VLAN tag on the phones directly.
   
  DV
  
  
_
  
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Alan 
  Neville
  Sent: Thursday, May 25, 2006 12:44 PM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] VLAN info
  
  
  
  Hello,
  
  I'm looking for any information on setting up VLANs to seperate the 
  telephony network from the ordinary network. I have google'd around 
  but haven't found a lot of information in the best way to go about 
  this. Has anyone managed to do this successfully in 
 conjunction with 
  asterisks? If so, could they provide an overview of what 
 they did and 
  how they find it? Did the performance improve after the VLANs were 
  setup?
  
  Many Thanks,
  
  
  Alan Neville
  
  Barlan Technologies,
  Technical Support and Helpdesk,
  [EMAIL PROTECTED]
  +353 1 866 6111
  
  
  
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 __
 _
 Faites de Yahoo! votre page d'accueil sur le web pour 
 retrouver directement vos services préférés : vérifiez vos 
 nouveaux mails, lancez vos recherches et suivez l'actualité 
 en temps réel. 
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Re: [Asterisk-Users] Is NuFone Really Dead? Same as voipjet

[Peter Bowyer]

On 25/05/06, Kerry Garrison [EMAIL PROTECTED] wrote:


Just because their email address is [EMAIL PROTECTED] doesn't mean its
fast, or is even answered. It should be /dev/[EMAIL PROTECTED]


I agree. Others seem to rave about them, but I've had no luck
attracting their 'fast' support staff's attention, despite many emails
and direct followups to their promotional postings on the -biz list.

Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Is NuFone Really Dead? Same as voipjet

[Matt]

That's because their fast support staff is John

On 5/25/06, Peter Bowyer [EMAIL PROTECTED] wrote:

On 25/05/06, Kerry Garrison [EMAIL PROTECTED] wrote:

 Just because their email address is [EMAIL PROTECTED] doesn't mean its
 fast, or is even answered. It should be /dev/[EMAIL PROTECTED]

I agree. Others seem to rave about them, but I've had no luck
attracting their 'fast' support staff's attention, despite many emails
and direct followups to their promotional postings on the -biz list.

Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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[Asterisk-Users] Failover Problem

[Douglas Garstang]

I have a weird situation. A polycom phone is configured to use system pbx1 as 
the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three 
systems have identical sip.conf files. The phone is registered on pbx1.

I shut down the Asterisk application on pbx1. I make a call. The phone sends an 
INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The 
phone sends an ACK followed by the INVITE with the credentials. What does 
Asterisk do? It sends the 407 Proxy Auth message AGAIN. Why? The phone of 
course re-sends the INVITE with the credentials again, and of course Asterisk 
sends the 407 again. WHY?

It looks like an Asterisk problem. As far as I can see, the phone is doing the 
right thing. Asterisk should not be sending Proxy Auth required over and over 
again.

This problem completely invalidates any sort of Asterisk redundancy. If this 
doesn't work, Asterisk can't be used in a redundant configuration, and that's a 
deal breaker.

Has anyone reading this done this before?

Doug.
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Re: [Asterisk-Users] Is NuFone Really Dead?

[Andrew Kohlsmith]

On Thursday 25 May 2006 11:18, Carlos Chavez wrote:
  DOne that several times and I keep getting the same message.  Support
 never answers the phone or email so how can I tell them it is not
 working?

Do you have ticket numbers from the autoresponder on [EMAIL PROTECTED]

-A.
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Re: [Asterisk-Users] Misdn 0.2.1 BUSY tone

[Christophorus Laube]

 I have this problem on misdn 0.2.1:
 in extension.conf  i have such a situation;

 [misdn_incoming]
 exten = 06786541,1,Dial(SIP/203)

 where SIP/203 is a GXP-2000.

 I want to make the 203 to answer just one call at the same time, so i've
 disabled the call waiting feature on the phone, but when I do this the
 caller does not hear the Busy tone, it receives the telco Network
 error tone.
Don't you have to set up a context for the SIP-phone? In sip.conf where you 
defined the SIP peer you have to set something like context=my_sip. This is 
what you also have to set up in your extensions.conf

[misdn_incoming]
exten = 06786541,1,Dial(SIP/203)

[my_sip]
exten = _X.-Busy,1,Busy()

just have a look at show application busy on the CLI. 
You also have to set up the right indications in indications.conf. Just set 
the country value to the right value. Maybe even that will just solve your 
problem. 

By the way there is a much newer version of misdn available.


 I want the caller to receive the busy tone when the called is busy . How
 can I do this?
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Re: [Asterisk-Users] Failover Problem

[Kevin P. Fleming]

Douglas Garstang wrote:

 I shut down the Asterisk application on pbx1. I make a call. The phone sends 
 an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The 
 phone sends an ACK followed by the INVITE with the credentials. What does 
 Asterisk do? It sends the 407 Proxy Auth message AGAIN. Why? The phone of 
 course re-sends the INVITE with the credentials again, and of course Asterisk 
 sends the 407 again. WHY?

Have you opened a bug in the bug tracker with a proper SIP/debug trace
for this problem so that people who understand the code can try to help
you? Obviously this is not expected behavior.
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RE: [Asterisk-Users] What and When is the next version of Asterisk?

[Douglas Garstang]

 -Original Message-
 From: Sean Cook [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, May 24, 2006 5:36 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] What and When is the next version of
 Asterisk?
 
 
 Not necessarily... my understanding is that the feature 
 freeze was done
 about 2 months ago for 1.4 and the release cycle is 6 months 
 putting 1.4
 due for release here pretty soon...
 
 As to what is in the new release... probably most of the sip changes
 that Olle has been working on as well as the core module
 loader/unloader... beyond that... I don't know.

Those SIP changes scare the poop outta me. Relying on the usual scarce 
documentation, it seemed like the proposed SIP changes are going to break a lot 
of things including SIP peering. I saw posts from others concerned about this 
as well.

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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 22, Issue 132

[Dan Elder]

I haven't intentionally done anything to the CID, the odd thing is that
another aastra phone (480e) connected to the same port DOES display CID,
it's only these PT350s that can't seem to read it.. Been talking to Aastra
support  they've suggested some avenues to explore, but none of them have
worked... The PT350s definitely have a problem w/*'s cid signals.. Just not
sure where to look to track it down.

Thanks for your input! I'm still trying to get this darn issue sorted, as I
have boxes of these phones that I really want to deploy, but I need this CID
display for them to be usefull in our environment.

Thanks again!

Dan


Are you doing something funny with the CID on it's way to the phone? 
I've got a somewhat similar problem with an Aastra IP phone (yes, I did say
IP): it would NOT ring if the caller id 
started with an #. Maybe your Aastra PSTN phone got some of the same
(buggy?) handling of CID's?

Dan Elder wrote:
 Hi All, posted last week but didn't get any responses. I'm trying to 
 figure out why some of our analog phones aren't showing CID when 
 hooked up to asterisk. To recap, I have an Aastra Powertouch 350, 
 which shows caller ID fine when connected to the PSTN, but when hooked 
 up to asterisk, CID does not show. I've hooked up another phone to the 
 same * port that the Aastra phone is on,  it DOES show CID, so I'm 
 assuming my settings  such are at least partially correct, can anyone 
 point me to some options or areas I can look to troubleshoot this 
 issue? Been pulling my hair out on this for days  just can't seem to get
it sorted.

 I'm using asterisk 1.2.0 with a Carrier Access ABII channel bank. When 
 another CID capable phone is hooke up to the same port, CID works 
 fine, the Aastra phone is however unable to read the incoming CID from *
apparently.

 Any pointers would be greatly appreciated, I've searched the Wiki  
 the CID faq's to no avail.

 Thanks in advance

 Dan

 

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[Asterisk-Users] PCI Problems

[Sean Cook]

OK... maybe I got a little anxious and ran out and bought a Tyan GX28
with dual Opteron (dual core) processors.  (It is a nice server ;) )  I
did neglect to find out that you can not manually set the IRQ's on this
motherboard.   I am now stuck sharing an IRQ with the ethernet
controller and no foreseeable end to my dilemma. 

I have a Digium TE210P and zttest consistently runs at 99.97% which as
you guessed, is giving rather unpleasing sound quality.  My options as I
see it are:

1.  Buy a new server
2.  Buy a sangoma A102U

I am looking for practical suggestions from those of you out there who
have had a similar experience that may aid me in making this decision. 

Thank you,

Sean
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[Asterisk-Users] Re: DHCP configuration for Cisco 7960?

[Julian C. Dunn]

On Wed, 2006-05-24 at 15:25 -0400, Julian Dunn wrote:

 I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on
 FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on
 the wire. I wonder if anyone has done this before and therefore can
 validate whether or not the traffic I am seeing is normal.

Never mind this... the root cause turned out to be network problems (ARP
replies were not getting back to the phone). The phone has now been
successfully flashed  registers as a SIP device with my Asterisk
server.

- Julian

-- 
Julian C. Dunn, P.Eng.
Systems Administrator

e: [EMAIL PROTECTED]
p: 416-363-6316 x292
f: 416-363-6102

Devlin eBusiness Architects Inc.
185 Frederick Street
Toronto, ON M5A 4L4

1700-1050 West Pender Street
Vancouver, BC V6E 4T3
604-707-4000
1-877-612-4400

www.devlin.ca
www.decisionroom.com

PLAN  DESIGN  BUILD
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Re: [Asterisk-Users] PCI Problems

[Robert Webb]

Sean Cook wrote:

OK... maybe I got a little anxious and ran out and bought a Tyan GX28
with dual Opteron (dual core) processors.  (It is a nice server ;) )  I
did neglect to find out that you can not manually set the IRQ's on this
motherboard.   I am now stuck sharing an IRQ with the ethernet
controller and no foreseeable end to my dilemma. 


I have a Digium TE210P and zttest consistently runs at 99.97% which as
you guessed, is giving rather unpleasing sound quality.  My options as I
see it are:

1.  Buy a new server
2.  Buy a sangoma A102U

I am looking for practical suggestions from those of you out there who
have had a similar experience that may aid me in making this decision. 


Thank you,

Sean
___
  


Have you tried changing the PCI slot and resetting the bios config?

Robert
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RE: [Asterisk-Users] OT: AudioCodesMP124-C/FSX/AC/SIP

[William Piper]

Thank you for your help, however I still cannot get this thing to connect to
asterisk. I followed your instructions to the letter  couldn't get it to
connect.

I'd be happy to throw it up on a public IP if anyone would like to take a
crack at getting it to work.

Thanks,

bp

On 5/24/06, The VoIP Connection [EMAIL PROTECTED] wrote:
 Here are the step by step instructions for setting up a brand new
Audiocodes
 FXS gateway for use with an Asterisk server:

 Connect the gateway to a network switch and connect a computer to the same
 switch. Then configure the IP address of the computer to 10.1.10.2. Then
run
 your web browser and point it to http://10.1.10.10 and login using the
 information below.

 Default IP address: 10.1.10.10
 Default user name: Admin
 Default password: Admin

 Goto Quick Setup and change the following:
 IP Address = Set to the new IP address of the AudioCodes gateway
 Subnet Mask = Set to the correct netmask for your local network
 Default Gateway Address = Set to the correct gateway IP address for your
 local network
 Working With Proxy = Set to Yes
 Proxy IP Address = Set to the IP address of the Asterisk server
 Enable Registration = Set to Enable

 Restart the gateway then log back in using the new IP address.

 Goto Protocol Management - Protocol Definition - Proxy  Registration
 Registrar IP Address = Set to the IP address of the Asterisk server
 Registration Time = Set to 60
 Subscription Mode = Set to Per Endpoint
 Authentication Mode = Set to Per Endpoint

 Goto Protocol Management - Protocol Definition - DTMF  Dialing
 Max Digits In Phone Num = Set to a large enough number such as 32

 Goto Protocol Management - Protocol Definition - Coders
 Add coders as needed You need to set at least G.711U-law

 Goto Protocol Management - Endpoint Settings - Authentication
 Set SIP username and password for each port

 Goto Protocol Management - Endpoint Phone Numbers
 Enter an extension (phone) number for every used channel

 Your AudioCodes gateway is now ready.

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

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[Asterisk-Users] Dlink DG-104S

[William Piper]

List,

I have a Dlink DG-104S that I got off of EBay. It is password protected 
the default passwords don't work. 

Does anyone know how to reset this box to defaults?

Thanks,

bp

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[Asterisk-Users] Asterisk and sysmask - anyone?

[Erick Perez]

Has anyone run asterisk with SYSMASK?

http://wims.unice.fr/sysmask/doc/

or maybe securing asterisk installations have another howto anywhere?

--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] macro-dial

[Mimmus]

Philippe,
I understand what you say...
I'd like to free myself from AMP/Freepbx because I feel 
better if I have only'vi-made' configuration files I can 
tweak.
I'd like also to have macro-dial entirely in the dialplan 
without AGI script but without losing call-forwarding, do-not-disturb, etc. 
functionalities.At this moment, I cleaned up a lot 
of things but still have dialparties.agi. I hope to 
thrash it in some future, when I will be able to rewrite all logic in the 
diaplan.

Thanks
Domenico


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Philippe 
  LindheimerSent: Thursday, May 25, 2006 4:44 PMTo: 
  asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users] 
  macro-dial
  
  Domenico,
  
  as I mentioned: "...and extensions are not necessarily what you think 
  they areeither." AMP/Freepbx 'virtualizes' extensions. The basic concept 
  is that there are users and then there are devices. A user can have multiple 
  devices. The default shipping mode provides the 'extensions' tab which ends up 
  creating a user with the sam extension number as the device that you assign 
  them. However, if you flip to 'deviceanduser' mode (see /etc/amportal.conf - 
  AMPEXTENSIONS=) you will see that you now can control users separate from 
  devices and you can assign multiple devices to a single user or you can make a 
  device adhoc allowing any user to login to the device and it becomes their 
  phone until they logout.
  
  So as I mentioned, it isn't that simple, it is the reason for all the 
  various callerid macros, dialparties.agi, etc. that is there. If you want more 
  detail, in addition to digging in as you have, you may want to move over to 
  the freepbx.org site and/or the IRC. If you want to get rid of dialparties, 
  maybe you can get the entire functionality into a dialplan format (and 
  probably improve performance) and then submit it back. But as you've probably 
  seen, dialparties itself is inegrally interwoven with macro-dial and the 
  various other interdependencies throughout the dial plan, astdb, etc.
  
  p
  From: 
"Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - 
Non-Commercial 
Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 
2006 10:21:46 +0200Subject: RE: [Asterisk-Users] 
macro-dialHi,I digged in dialparties.agi and found that apart 
from DND, hunt-group,status, etc its main function is looking up real 
device(s) for any user fromAstDB. In fact, AMP/FreePBX keep a long list 
of entries in AstDB for anydevice/user.I'm interested in knowing 
how people on this list manage link between anextension and the real 
device (SIP, Zap, 
etc).ThanksFrom: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED] 
On Behalf Of PhilippeLindheimerSent: Wednesday, May 24, 2006 7:42 
PMTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] 
macro-dialIt's not that simple. dialparties is fundamental to 
the wholedialplan in AMP/freepbx and accomplishes a lot of the features 
such as huntgroups, DND, etc. And extensions are not necessarily what 
you think they areeither. If you don't like it, you'd probably be better 
off writing your owndialplan or alternatively, rewrite it's entire 
functionality outside of anagi and then submit the mod to freepbx to 
streamline freepbx more.pFrom: "Mimmus" 
<[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial 
Discussion'"Date: Wed, 24 May 2006 
18:00:36 +0200Subject: [Asterisk-Users] macro-dialHi,I'm 
trying to edit an AMP-derived dialplan: the macro "dial" usesthe 
AGIscript "dialparties.agi" to find the extension to call.I'd like 
to drop this script: does anyone can explain me what is 
itsmainjob?Thanks-- Domenico 
  Viggiani
  
  
  Blab-away for as little as 1¢/min. Make PC-to-Phone 
  Calls using Yahoo! Messenger with Voice.
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[Asterisk-Users] Asterisk Manuals

[Erick Perez]

A customer is asking for a manual. He's not talking about a How-To.
He's talking about a PDF/DOC that shows what files do what and what
parameters can be used and the syntaxis.

I think Asterisk Business Edition has one that comes with the box.

For the recent *not ABE* stable release will be nice.

Comments?


--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] Problems with Park and MOH

[Doug Lytle]

Bob McDowell wrote:
Can I ask how you set the parking lots to a different moh?  I don't 
see a setting for it in features.conf...


All incoming lines are MP3, all others are Native MOH.  I use the 
SetMuiscOnHold option though out the dial plan.  Tape being Native and 
cd being mp3.


musiconhold.conf

[tape]
mode=files
directory=/var/lib/asterisk/moh-native/tape

[cd]
mode=mp3
directory=/var/lib/asterisk/mohmp3/epi-video


zapata.conf


[channels]
context=incoming
callerid=Outside (734) xxx-
musiconhold=tape
channel = 1

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Re: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501phone

[Doug Lytle]

[EMAIL PROTECTED] wrote:

Could not find your post for 4 months ago.


Anton's post from March:

| Man! I love polycoms.. They are good phones and highly configurable. 


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|[EMAIL PROTECTED]

|Sent: Tuesday, March 07, 2006 7:41 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Polycom 
|voice.gain.tx.analog.handsetandasteriskecho

|
||While I'm asking about the Polycom ip500, the answers for all phones 
||where mic/handset/headset levels are adjustable would be of

|interest to
||many I'm sure.
||
||For the ip500, the default value for the handset seems to be 
||voice.gain.tx.analog.handset=3

|
|I have a number of IP600s and 601s that I was experiencing 
|occassional echo with.  I recently upgraded them to firmware 
|1.6.5, and rather than using my existing sip.cfg/ipmid.cfg 
|that had been around forever I started fresh with a completely 
|stock 1.6.5 sip.cfg file.  My echo issues have disappeared completely.

|
|With the 1.6.5 version of the Polycom firmware the default 
|value for voice.gain.tx.analog.handset=12.  The default 
|value for voice.gain.tx.analog.headset=3.  I suspect you 
|should update the entire voice section of the file (if 
|you're not ready to start from scratch) since it contains 
|default values for AEC, AES, NS, AGC, RXEQ, and TXEQ.  I have 
|pasted just the gains section below in case anyone want to 
|compare it to their current settings.

|
|  gains
| voice.gain.rx.analog.handset=0
| voice.gain.rx.analog.headset=0
| voice.gain.rx.analog.chassis=0
| voice.gain.rx.analog.chassis.IP_300=-6
| voice.gain.rx.analog.chassis.IP_4000=3
| voice.gain.rx.analog.chassis.IP_601=6
| voice.gain.rx.analog.ringer=0
| voice.gain.rx.analog.ringer.IP_300=-6
| voice.gain.rx.analog.ringer.IP_4000=3
| voice.gain.rx.analog.ringer.IP_601=6
| voice.gain.rx.digital.handset=-15
| voice.gain.rx.digital.headset=-21
| voice.gain.rx.digital.chassis=0
| voice.gain.rx.digital.chassis.IP_4000=0
| voice.gain.rx.digital.chassis.IP_601=0
| voice.gain.rx.digital.ringer=-21
| voice.gain.rx.digital.ringer.IP_4000=-21
| voice.gain.rx.digital.ringer.IP_601=-21
| voice.gain.rx.analog.handset.sidetone=-14
| voice.gain.rx.analog.headset.sidetone=-24
| voice.gain.tx.analog.handset=12
| voice.gain.tx.analog.headset=3
| voice.gain.tx.analog.chassis=3
| voice.gain.tx.analog.chassis.IP_300=0
| voice.gain.tx.analog.chassis.IP_4000=3
| voice.gain.tx.analog.chassis.IP_601=0
| voice.gain.tx.digital.handset=0
| voice.gain.tx.digital.headset=0
| voice.gain.tx.digital.chassis=3
| voice.gain.tx.digital.chassis.IP_4000=0
| voice.gain.tx.digital.chassis.IP_601=6
| voice.gain.tx.analog.preamp.handset=14
| voice.gain.tx.analog.preamp.headset=23
| voice.gain.tx.analog.preamp.chassis=32
| voice.gain.tx.analog.preamp.chassis.IP_601=32/
|



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Re: [Asterisk-Users] Asterisk Manuals

[Samy Antoun]

--- Erick Perez [EMAIL PROTECTED] wrote:

 A customer is asking for a manual. He's not talking about a How-To.
 He's talking about a PDF/DOC that shows what files do what and what
 parameters can be used and the syntaxis.
 
 I think Asterisk Business Edition has one that comes with the box.
 
 For the recent *not ABE* stable release will be nice.
 
 Comments?

Eric,

Here is a very good set of documents:
http://www.asterisk.org/doxygen/



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RE: [Asterisk-Users] Failover Problem

[Douglas Garstang]

 -Original Message-
 From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
 Sent: Thursday, May 25, 2006 9:43 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Failover Problem
 
 
 Douglas Garstang wrote:
 
  I shut down the Asterisk application on pbx1. I make a 
 call. The phone sends an INVITE to pbx2. Pbx2 sends back a 
 407 Proxy Auth message to the phone. The phone sends an ACK 
 followed by the INVITE with the credentials. What does 
 Asterisk do? It sends the 407 Proxy Auth message AGAIN. Why? 
 The phone of course re-sends the INVITE with the credentials 
 again, and of course Asterisk sends the 407 again. WHY?
 
 Have you opened a bug in the bug tracker with a proper SIP/debug trace
 for this problem so that people who understand the code can 
 try to help
 you? Obviously this is not expected behavior.
I have now...
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Re: [Asterisk-Users] PCI Problems

[Rob Lith]

Does the sangoma handle sharing interrupts in some other way?RobOn 25/05/06, Sean Cook [EMAIL PROTECTED]
 wrote:OK... maybe I got a little anxious and ran out and bought a Tyan GX28with dual Opteron (dual core) processors.(It is a nice server ;) )I
did neglect to find out that you can not manually set the IRQ's on thismotherboard. I am now stuck sharing an IRQ with the ethernetcontroller and no foreseeable end to my dilemma.I have a Digium TE210P and zttest consistently runs at 
99.97% which asyou guessed, is giving rather unpleasing sound quality.My options as Isee it are:1.Buy a new server2.Buy a sangoma A102UI am looking for practical suggestions from those of you out there who
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RE: [Asterisk-Users] Failover Problem

[Douglas Garstang]

I just realised that in actual fact Asterisk is displaying on the console 
'Ignoring this INVITE request'. 
Any ideas why it would be doing that? It doesn't say WHY...

 -Original Message-
 From: Douglas Garstang 
 Sent: Thursday, May 25, 2006 10:41 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Failover Problem
 
 
  -Original Message-
  From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
  Sent: Thursday, May 25, 2006 9:43 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Failover Problem
  
  
  Douglas Garstang wrote:
  
   I shut down the Asterisk application on pbx1. I make a 
  call. The phone sends an INVITE to pbx2. Pbx2 sends back a 
  407 Proxy Auth message to the phone. The phone sends an ACK 
  followed by the INVITE with the credentials. What does 
  Asterisk do? It sends the 407 Proxy Auth message AGAIN. Why? 
  The phone of course re-sends the INVITE with the credentials 
  again, and of course Asterisk sends the 407 again. WHY?
  
  Have you opened a bug in the bug tracker with a proper 
 SIP/debug trace
  for this problem so that people who understand the code can 
  try to help
  you? Obviously this is not expected behavior.
 I have now...
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Re: [Asterisk-Users] IVR transcoding g729 license

[Tamas]

Kristian Kielhofner wrote:
 Tamas wrote:
 Hello,

 - a call comes in with protocol SIP, codec g729
 - connects to the IVR [playing prompts, collecting digits]
 - dialing forward to protocol IAX2, codec g729

 I guess, while the call is in IVR, I need one g729 license for the call.
 If the call leaves the IVR, thus dials to another box with the same g729
 codec, will it need 2 licenses? While there should be no transcoding
 further, will it free up the codec which was used during IVR session?

 I would like to avoid useless transcoding [g729-slin-g729] in case the
 call goes through an IVR.

 How is such case handled in asterisk? [1.2]

 Thanks in advance,
 Tamas

 Tamas,

 Just record your IVR prompts in g729 (or convert them to g729 from
 some other format)!

 -- 
 Kristian Kielhofner

Hello,

that's a good idea! Would it mean, I don't even need a g729 license?
Thanks!

Tamas


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[Asterisk-Users] Asterisk codec negotiation patch

[Erick Perez]

does anybody knows if this patch made it into Asterisk Business Edition?
http://bugs.digium.com/view.php?id=4825

The latest release is :
asterisk127_codec_negotiation-20060505.diff.gz

We are unsure as to buy ABE or go and download the open source
version, in terms of patch availability and features.

Comments are welcomed,

--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] PCI Problems

[Sean Cook]

Rob Lith wrote:
 Does the sangoma handle sharing interrupts in some other way?
from:  http://www.voip-info.org/wiki/view/Sangoma

There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
hardware and ANY make/brand of PC/server- NONE

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Re: [Asterisk-Users] Asterisk codec negotiation patch

[Kevin P. Fleming]

Erick Perez wrote:
 does anybody knows if this patch made it into Asterisk Business Edition?
 http://bugs.digium.com/view.php?id=4825

ABE never includes any features that are not in open source Asterisk,
except for things that cannot be done via an open source license.

No patches from Mantis will ever be in ABE before they are merged in
open source Asterisk.
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[Asterisk-Users] Failover Problem

[Brent Torrenga]

Doug,

I think I have seen this - does the initial invite to pbx2 make complete
sense - is it valid according to the sip.conf entry on pbx2? In the pbx2
sip.conf, try insecure=invite, maybe insecure=invite,port/insecure=port. I
don't have a real handle on why, but I recall it solving some sort of
similar behavior I saw, though it was in a different situation.

I think I had 2 * servers talking to each other like your phone and pbx2:
when someone tried to dial a sip URI to the second asterisk server using the
first asterisk server as the outgoing proxy.

I have a weird situation. A polycom phone is configured to use system pbx1
as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All
three systems have identical sip.conf files. The phone is registered on
pbx1.

I shut down the Asterisk application on pbx1. I make a call. The phone
sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the
phone. The phone sends an ACK followed by the INVITE with the credentials.
What does Asterisk do? It sends the 407 Proxy Auth message AGAIN. Why? The
phone of course re-sends the INVITE with the credentials again, and of
course Asterisk sends the 407 again. WHY?

It looks like an Asterisk problem. As far as I can see, the phone is doing
the right thing. Asterisk should not be sending Proxy Auth required over
and over again.

This problem completely invalidates any sort of Asterisk redundancy. If
this doesn't work, Asterisk can't be used in a redundant configuration, and
that's a deal breaker.

Has anyone reading this done this before?

Doug.


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

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

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
www.torrenga.com

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Re: [Asterisk-Users] Asterisk codec negotiation patch

[Erick Perez]

When things from MANTIS are merged into stable asterisk. Where does it
says that?


On 5/25/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:

Erick Perez wrote:
 does anybody knows if this patch made it into Asterisk Business Edition?
 http://bugs.digium.com/view.php?id=4825

ABE never includes any features that are not in open source Asterisk,
except for things that cannot be done via an open source license.

No patches from Mantis will ever be in ABE before they are merged in
open source Asterisk.
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--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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[Asterisk-Users] Compilation issues with s390

[Frank Pani]

Hi all,

I'm trying to compile asterisk on the mainframe (s390 / s390x) and I am
running into issues.   I was wondering if somebody could give a hand?

I'm thinking that I should be able to do this.  I have noticed that Debian
even has binary RPM's out for Asterisk now.   I'm trying to do this on SuSE
SLES8 (with the 2.4 kernel).

What I see is, an issue that arch=s390 isn't supported and I wonder - is
there a way around this to make asterisk to compile. Again, I'm thinking
that this has to work.

 compilation runs successfully up to this point:

gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1
-funroll-loops -O6 -march=s390 -fPIC -DSASR -DNDEBUG -DWAV49 -I./inc
src/add.c
cc1: invalid option `arch=s390'
make[2]: *** [src/add.o] Error 1
make[2]: Leaving directory `/usr/src/asterisk-1.2.7.1/codecs/gsm'
make[1]: *** [gsm/lib/libgsm.a] Error 2
make[1]: Leaving directory `/usr/src/asterisk-1.2.7.1/codecs'
make: *** [subdirs] Error 1

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Re: [Asterisk-Users] US telco lingo

[Don Pobanz]

Brian C. Fertig wrote:

I think dude was trying to be a smart ass or show us his experience in
telecom..  :)  At least he knows the pinout for a T1..   



I have been properly put in my place by you and many others..  :) After 
rereading the original post, I don't believe it has anything to do with 
jacks. I use to work for a local phone company where we regularly did T1 
installs and the only 48 we used was part of a rj48 jack. Thanks, for 
not letting anything foolish get through!!! :)


Don Pobanz


-Original Message-n to a non-US dummy the following phrases I have

What is US48?


I assume by US48 they mean RJ48 which is a 8 conductor modular jack

with

signal from the phone company on 12 and signal to the phone company

on

45.

Don Pobanz


You are kidding right???



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RE: [Asterisk-Users] Asterisk codec negotiation patch

[Alexander Lopez]

You can look at the change logs

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Erick Perez
 Sent: Thursday, May 25, 2006 1:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk codec negotiation patch
 
 When things from MANTIS are merged into stable asterisk. Where does it
 says that?
 
 
 On 5/25/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
  Erick Perez wrote:
   does anybody knows if this patch made it into Asterisk Business
 Edition?
   http://bugs.digium.com/view.php?id=4825
 
  ABE never includes any features that are not in open source
Asterisk,
  except for things that cannot be done via an open source license.
 
  No patches from Mantis will ever be in ABE before they are merged in
  open source Asterisk.
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 --
 
 ---
 Erick Perez
 Linux User 376588
 http://counter.li.org/  (Get counted!!!)
 Panama, Republic of Panama
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RE: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

I understand, seems like it might be easier to write a new dialplan from scratch though, vs. running into all sorts of strange issues? On the other hand, doing it your way will make you understand what freepbx is doing, which migh provide for your own ideas on how to do or not to do things in your own dialplan.p  From: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 18:25:15 +0200Subject: RE: [Asterisk-Users] macro-dialPhilippe,  I understand what you say...  I'd like to free myself from AMP/Freepbx because I feel better if I have only'vi-made' configuration files I can tweak.  I'd like also to have macro-dial entirely in the dialplan without AGI script but without losing call-forwarding, do-not-disturb, etc. functionalities.At this moment, I cleaned up a lot of things but still have dialparties.agi. I hope to thrash it in some future, when I will be able to rewrite all logic in the diaplan.Thanks  Domenico  
		Be a chatter box. Enjoy free PC-to-PC calls  with Yahoo! Messenger with Voice.___
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RE: [Asterisk-Users] US telco lingo

[Brian C. Fertig]

Well we try..  


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz
Sent: Thursday, May 25, 2006 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] US telco lingo

Brian C. Fertig wrote:
 I think dude was trying to be a smart ass or show us his experience in
 telecom..  :)  At least he knows the pinout for a T1..   
 

I have been properly put in my place by you and many others..  :) After 
rereading the original post, I don't believe it has anything to do with 
jacks. I use to work for a local phone company where we regularly did T1

installs and the only 48 we used was part of a rj48 jack. Thanks, for 
not letting anything foolish get through!!! :)

Don Pobanz

 -Original Message-n to a non-US dummy the following phrases I
have
 What is US48?

 I assume by US48 they mean RJ48 which is a 8 conductor modular jack
 with
 signal from the phone company on 12 and signal to the phone company
 on
 45.

 Don Pobanz
 
 You are kidding right???
 

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Re: [Asterisk-Users] Asterisk codec negotiation patch

[Kevin P. Fleming]

Erick Perez wrote:
 When things from MANTIS are merged into stable asterisk. Where does it
 says that?

In Mantis, and in the commit message for the Subversion repository.

But the patch you are asking about will not be merged into the 1.2.x
release branch (which we no longer call 'stable Asterisk'), since it is
a new feature. It hasn't even been decided whether it will get merged
into the development branch for 1.4, although that decision will get
made very soon.
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Re: [Asterisk-Users] Compilation issues with s390

[Kevin P. Fleming]

Frank Pani wrote:

  compilation runs successfully up to this point:
 
 gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
 -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
 -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1
 -funroll-loops -O6 -march=s390 -fPIC -DSASR -DNDEBUG -DWAV49 -I./inc
 src/add.c
 cc1: invalid option `arch=s390'
 make[2]: *** [src/add.o] Error 1
 make[2]: Leaving directory `/usr/src/asterisk-1.2.7.1/codecs/gsm'
 make[1]: *** [gsm/lib/libgsm.a] Error 2
 make[1]: Leaving directory `/usr/src/asterisk-1.2.7.1/codecs'
 make: *** [subdirs] Error 1

This happens frequently with the Makefile for the embedded GSM library.
I will add a filter to stop that from happening that will be in the next
release, and if you watch the asterisk-commits mailing list you can
extract the (very small) patch and apply it manually on your system (or
pull down the SVN branch-1.2 code instead of using the 1.2.7.1 tarball).
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Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750

[Benchev]

Hi Nguyen ,
I haven't got the opportunity to make my project real due to business
obstacles, but I still think that it should work.
All that follows is a theory, but there are guys on the list that
might help you with more practical advises.
 I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the
 manual of Hipath 3500 yet (have to buy from local vendor), so I was not
 sure are these thing possible

 Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTN
I had the same idea because I wanted to save on the card side(single span), 
and use  the Hipath as a channel  bank :-)

 - Is this possible for Asterisk Users call out using CO lines? Some of
 Siemens guys told me that I need an DISA card for this? Is this true?
Most of the time the Siemens guys don't know what is Asterisk.
Basically TE110P *is* a DISA since it gives Direct Inward System Access
(if this is what they mean by DISA)

Below is a threat I found with exactly the same scenario like yours:
http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html
And this proves that the idea must work.
 - When the call arrived from PSTN through CO line, can it be forwarded to
 Asterisk? Again, they says that we require the DISA card.

As far as anything gets into Asterisk then you are free to do whatever you
want. I don't know what DISA they are talking about? Do they mean S2M
or similar thing(but TMS2 is S2M)?
 Anyone?

Sorry for not being able to help, but hope somebody else
would do it.

Benchev


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[Asterisk-Users] FreePBX virtualization

[Daniel Salama]

Does FreePBX support virtualization of its services? For example, can  
I use it to provide virtual PBX to different clients under the same  
instance of FreePBX? Or is it more geared to single office-type  
installation?


Thanks,
Daniel
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Re: [Asterisk-Users] Asterisk codec negotiation patch

[Erick Perez]

many thanks to all.


On 5/25/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:

Erick Perez wrote:
 When things from MANTIS are merged into stable asterisk. Where does it
 says that?

In Mantis, and in the commit message for the Subversion repository.

But the patch you are asking about will not be merged into the 1.2.x
release branch (which we no longer call 'stable Asterisk'), since it is
a new feature. It hasn't even been decided whether it will get merged
into the development branch for 1.4, although that decision will get
made very soon.
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--

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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RE: [Asterisk-Users] FreePBX virtualization

[Kerry Garrison]

You can by creating different contexts and using the Administrators function
allow them to modify some of the settings themselves.
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Daniel Salama
 Sent: Thursday, May 25, 2006 10:42 AM
 To: Non-Commercial Discussion Asterisk
 Subject: [Asterisk-Users] FreePBX virtualization
 
 Does FreePBX support virtualization of its services? For 
 example, can I use it to provide virtual PBX to different 
 clients under the same instance of FreePBX? Or is it more 
 geared to single office-type installation?
 
 Thanks,
 Daniel
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Re: [Asterisk-Users] PCI Problems

[Andrew Kohlsmith]

On Thursday 25 May 2006 13:06, Sean Cook wrote:
 There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
 hardware and ANY make/brand of PC/server- NONE

Not exactly.  Their hardware and drivers play nicely far more often than the 
older Digium boards, but I have personally been bitten by their S518 not 
playing nice with their A101u in a Dell P3 system.  The S518's driver would 
cause very clear and repeatable audio chirping on the T1.  Replace the 
A101u with a T100P and no issues whatsoever.

This was about 18 months ago.  Drivers have changed since then, and this may 
no longer be an issue. That particular system has been EOLd so it's a 
complete non-issue to me.

I've mentioned this to Sangoma in passing, but seeing as it is a fairly 
irregular case, I don't expect any fuss to be made of it.  Digium's cards are 
historically much finickier (is that a word?) but I know that their rev2 
TE405/410 and latest TDM400 carrier is significantly better than their older 
stuff in terms of compatibility.

Technically speaking, both Sangoma and Digium use the *identical* Xilinx 
Spartan II FPGA for their PCI interface (on the A104 and TE405/410).  I feel 
fairly confident to also say that they will be using the same PCI VHDL 
block to interface that part to the PCI bus.  So what this comes down to, 
by and large, is the drivers.

There have been some hardware PCI interop issues with Digium's stuff, but I 
know for a fact that these have been fixed in their rev2 hardware.

Also technically speaking (but not PCI-speaking), Sangoma's multiport cards 
can do some things that the Digium cards cannot.  Specifically, Sangoma's 
cards have no issues whatsoever with having their spans synchronized to 
completely different clocking sources.  This is achieved by using a 
single-port framer for each span.  Digium's framer is a single chip that 
supports four spans, and one of the framer's limitations is that all spans of 
the same technology (T1/E1) must share a clocking source.  (I've dug around 
in the framer's datasheet several months ago for a separate nefarious 
project, which is why I know this.)  For MOST people this isn't an issue, but 
it comes up now and again.

Sangoma's single, dual and quadspan cards can also fit in a half-height PCI 
slot.  Digium's quadspan cannot, and I believe that their dualspan cannot do 
this, either.  For some people this is an issue.

Sangoma's echo cancellation option board also seems to be far heftier than 
Digium's VPM.  I've got an A104d that replaced a TE406 and completely 
eradicated all of my echo troubles.

I like Digium's products, and I like Sangoma's products.  I'm not endorsing 
one over the other in this post at all; I'm merely pointing out that your 
none and never are strong words that no vendor can meet.  Sangoma's 
response to oddball hardware has been MUCH more customer-oriented than 
Digium's, at least historically speaking.  I believe that Digium's made 
*significant* improvements in their customer relations over the past little 
while, but by and large the products I buy from both vendors just works and 
I hardly ever have to call support.

-A.
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Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750

[Josué Conti]

Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk(
1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk.

I wait to have helped.
Greetings
Josué


2006/5/25, Benchev [EMAIL PROTECTED]:
Hi Nguyen ,I haven't got the opportunity to make my project real due to businessobstacles, but I still think that it should work.
All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not
 sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-)
 - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk.
Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours:
http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card.
As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else
would do it.Benchev___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list
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Re: [Asterisk-Users] FreePBX virtualization

[Daniel Salama]

Not really looking to give the client web access. Just trying to  
make my life easier :)


Thanks,
Daniel

On May 25, 2006, at 2:07 PM, Kerry Garrison wrote:

You can by creating different contexts and using the Administrators  
function

allow them to modify some of the settings themselves.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: Thursday, May 25, 2006 10:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] FreePBX virtualization

Does FreePBX support virtualization of its services? For
example, can I use it to provide virtual PBX to different
clients under the same instance of FreePBX? Or is it more
geared to single office-type installation?

Thanks,
Daniel
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