Re: [asterisk-users] How to integrate freepbx with a2billing?

[Sharon Lim]

I have trixbox installed but dont see a2billing installed together with it...anyone have integrate this before or is there any billing system that can integrate with freepbx. thanks
On 9/12/06, William Piper [EMAIL PROTECTED] wrote:
Both trixbox and asterisk2billing have their own lists... you may have better luck searching there.

bp
On 9/11/06, Steve Totaro [EMAIL PROTECTED]
 wrote:
Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see
 both application is not integrated special on cdr part. Any idea how to integrated it?Confuse! -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back
 pocket *You could just install TrixBox and disable a couple things, then youwould have a working system or one you could at least look at how it isconfigured to get ideas on how to configure your machine.
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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Re: [Asterisk-Users] Sirrix BRI errors

[Klaus Darilion]

[EMAIL PROTECTED] wrote:

Hi

I have a test setup of a sirrix card installed in NT mode connected to a
PBX.  I keep getting the following error:

   D-Channel receive message aborted, discarding frame (RSTAD=0x1c)

What does this mean?  What could be causing it?


The answer comes a little bit late, but still good for the archives:

You have to activate termination of the ISDN line (activate the jumpers 
on the Sirrix card).


regards
klaus
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[asterisk-users] Re: MSSQL connection

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi everyone,
 
 I am looking to log CDR records to our MSSQL database for further 
 examination on the records. From what I gathered from the wiki I have to 
 choose between FreeTDS and unixODBC. Is there a better choice? Which 
 option would be better in the log run?

Hi Kevin!

I'm using unixODBC drivers for storing cdr data to MSSQL 2000 database. It's 
working on two different systems.

Your problem could be that in database you have defined domain user instead of 
system user. And you have granted rights for that domain user. Anyway, that was 
my problem. Maybe you have make the same mistake :))

One thing about unixODBC. I'm looking for information how long does unixODBC 
holds data if it's unable to send it to MSSQL server? 


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[Asterisk-Users] Browsing distant missed call list

[Olivier]

Hi,Are you aware of any SIP hardphone (or softphone) offering the option to browse its missed call list from a distant (xml, SQL or whatever) server instead of using its own list ?This would be very useful to avoid duplicate entries for instance, when an incoming call is forwarded from one extension to another.
Doing my homework, I could find http://www.voip-info.org/wiki/view/snom+360but this doesn't really answer the point.Cheers
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Re: [asterisk-users] Dell hardware ...

[Giorgio Incantalupo]

Hi Alan,
simply do not use Dell hardware.
we had your problem, we called Dell and they told us that our servers 
was not configurable (they were too cheap).

So now I do not use Dell anymore and we have less problem.


Giorgio Incantalupo


Alan Bunch wrote:
I was going to use a Dell 1425 for Asterisk build but I see on 
Digium's website that hardware may be problematic.  Can anyone shed a 
litle more light on the problem.   I see the Intel ethernet cards seem 
to cause problems.  If I need to disable the onboard Intel on the Dell 
hardware I can I just need to know what to expect.


How about the 850, any word there  ?

TIA

Alan




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[Asterisk-Users] The best way to design local-only off-hours ringing

[Olivier]

Hi,During off-hours, we often set Asterisk server to ring all extensions.Sometimes, some of these are diverted to mobile or off-site numbers.Which is the best way to handle this ie to make sure only not diverted extensions are ringed ?
My understanding is Asterisk cannot know in advance which extensions are diverted (as call diversion remains inside phone memory)I was thinking of either collecting 3xx and 180 messages or creating a specific context forbidding outside calls but I'm still wondering of a smarter way to do it.
What do you think ?Cheers
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[asterisk-users] Re: Take 3 -- Trying to get SIP firmware on a 7970G

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 The loadInformation line in the SEP file reads like this:
 loadInformationSIP70.8-0-4SR1S/loadInformation
 
 - Here are TFTP server logs to illustrate that I'm using the correct  
 case'd XmlDefault.cnf.xml file:
 
 Sep 10 21:57:55 bubbles  tftpd[89195]: jalc7970.sip : read request  
 for SEP00131A4D39F4.cnf.xml: File not found
 Sep 10 21:57:55 bubbles  tftpd[89197]: jalc7970.sip : read request  
 for //XmlDefault.cnf.xml: success
 
 - All the files from the .cop are 100% unmodified.  I just tar -zxvf  
 cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into  
 the tftpd root directory, which is the same place the SEP and  
 XmlDefault file are located.
 
 /stumped.

Hi Jason!

Try those two things:
- don't use 8.0.4 firmware, use 8.0.2
- I didn't get my firmware from cop file, I have downloaded zip file



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] about 'zap show channels'

[PSPunch]

Hi all.

I've got a question regarding usages with a TDM400.


In results from the command

$ asterisk -rx 'zap show channels [n]'

I've noticed a change that may have been made somewhere between
ver 1.2.9.1 and 1.2.11.

In the older versions, the line Real: seemed to contain the text
'Linear' if the person on the zap channel was on hold.

This feature seems to have been eliminated in 1.2.11.
Now the results of the above command seem identicle no matter the
zap line is put on hold or not.

I used to rely on this fact when checking weather a person on the
zap channel was on hold or not with scripts and am facing some
inconveniences now.


I was hoping someone could give me some input on

1. What the text Linear really meant.
2. If the feature is gone, if so, for what reason.
3. Any other work arounds on checking if a zap channel is currently
   on hold from the command line.


Thank you.


--
David Shimamoto
[EMAIL PROTECTED]
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[asterisk-users] SIP/2.0 403 Relaying denied

[Rene]

Hi all,

I am trying to register myself to my VOIP provider (budgetphone.nl) so I
can accept inbound calls. However, using sip debug I get the following
error:

-- SIP read from 81.23.228.150:5060:
SIP/2.0 403 Relaying denied
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK65bc02f0;rport=1048
From: asterisk sip:[EMAIL PROTECTED];tag=as67980981
To: sip:sip.budgetphone.nl;tag=48cd709ae35c71b0e4d33846252f55ea.b365
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Server: OpenSer (1.0.2-tls (i386/linux))
Content-Length: 0

I am behind a NAT device which forwards port 5060 to my asterisk server.
Outgoing calls do work OK.

This is what I have in my sip.conf:
register = 3172xx:[EMAIL PROTECTED]

When I do sip show registry I do not see anything registered.
Does someone know what SIP/2.0 403 Relaying denied means?

Thanks,
Rene

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[asterisk-users] asterisk logging per day

[Christophorus Laube]

Hi list,

I am searching for a possibility to let my * log per day. So that a new
logfile is taken every night at midnight, with the date in the file name.
Is there a way to do so? Does anyone of you has tried that before?
Regards, Christophorus
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Re: [asterisk-users] question...

[Rich Adamson]

If you have four pstn telephone numbers (eg, 444-1212, 444-1213, 
444-1214, and 444-1215) from your telco, then call the telco and have 
them implement call forwarding on each of the four lines. You might also 
verify they provide a call forwarding on busy function for those lines.


After they have implemented it, put an analog phone on line 444-1212 and 
implement call forwarding on busy using whatever codes are appropriate 
(*90 here), forwarding calls to 444-1213. Do the same for 444-1213 and 
444-1214.


Now when 444-1212 is busy, the next incoming call goes to 444-1213. When 
444-1213 is busy, the next call goes to 444-1214, etc.



Christopher Corn wrote:

rich,
thanks for replying. i assume your talking about enabling call forward 
and call forward on busy from my vsp side. i dont quite grasp everything 
else that your saying, can you explain in laymen terms. thanks.


*/Rich Adamson [EMAIL PROTECTED]/* wrote:

Christopher Corn wrote:
  i plan on buying 4 residential lines for our small office and i was
  giving some thought. we'd like to have one main number that can
transfer
  calls to the other lines. but seeing that i have 4 different
individual
  lines with different numbers, im not seeing hows thats possible,
without
  tying up a line on the main phone. i would think i would need one DID
  with multiple simultaneous connections.

Two ways to accomplish the objective.

1. ask the telco about four lines in a trunk group (or sometimes
referred to as a rotary hunt group).

2. Subscribe to call forwarding on each line, and program each line for
call forward on busy to the next line of the four. It will accomplish
the same thing as the trunk group approach above.

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Re: [asterisk-users] asterisk logging per day

[Alberto Sagredo]

You could use logrotate or you could configure your cron to send 
asterisk -x logger rotate, which it will do what you want.


Regards



Christophorus Laube escribió:

Hi list,

I am searching for a possibility to let my * log per day. So that a new
logfile is taken every night at midnight, with the date in the file name.
Is there a way to do so? Does anyone of you has tried that before?
Regards, Christophorus
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--
Alberto Sagredo
I+D Area (Asterisk // Cisco-Linksys)
Peoplecall


Email : [EMAIL PROTECTED]
Blog: http://www.voipnovatos.es

Tel./Ph. : +34 91 120 5080
Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39
Fax./Fax.: +34 91 661 9460
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[asterisk-users] Samsung OfficeServ 500 + Asterisk(Tormenta 2) via PRI

[Eugeniy Khvastunov]

Kind time of day, All!
Prompt, please!
Is Samsung OfficeServ 500 in it card TEPRI is established, also there is 
a server on Gentoo with Asterisk PBX + Established card Tormenta 2 (4 
ports PRI). On Asterisk awakes it is submitted 3 PRI a stream from PSTN 
and from it in turn on Samsung OfficeServ 500 + are planned soft VoIP 
phones.
There is a configuration file for Tormenta 2 (zaptel.conf), actually now 
a question:

What signaling system PRI i need to use?:
# Next come the definitions for using the channels. The format is:
# device=channel list
#
# Valid devices are:
#
# em : Channel(s) are signalled using EM signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# fxsls : Channel(s) are signalled using FXS Loopstart protocol.
# fxsgs : Channel(s) are signalled using FXS Groundstart protocol.
# fxsks : Channel(s) are signalled using FXS Koolstart protocol.
# fxols : Channel(s) are signalled using FXO Loopstart protocol.
# fxogs : Channel(s) are signalled using FXO Groundstart protocol.
# fxoks : Channel(s) are signalled using FXO Koolstart protocol.
# sf : Channel(s) are signalled using in-band single freq tone.
# Syntax as follows:
# channel# = sf:rxfreq,rxbw,rxflag,txfreq,txlevel,txflag
# rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
# bandwith in hz (typically 10.0), rxflag is either 'normal' or
# 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
# level in dbm, txflag is either 'normal' or 'inverted'. Set
# rxfreq or txfreq to 0.0 if that tone is not desired.
# unused : No signalling is performed, each channel in the list 
remains idle

# clear : Channel(s) are bundled into a single span. No conversion or
# signalling is performed, and raw data is available on the master.
# indclear: Like clear except all channels are treated individually and
# are not bundled. bchan is an alias for this.
# rawhdlc : The zaptel driver performs HDLC encoding and decoding on the
# bundle, and the resulting data is communicated via the master
# device.
# fcshdlc : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoing FCS insertion
# and verification. dchan is an alias for this.
# nethdlc : The zaptel driver bundles the channels together into an
# hdlc network device, which in turn can be configured with
# sethdlc (available separately).
# dacs : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# dacsrbs : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon and
# also performs the DACSing of RBS bits

Thanks!
begin:vcard
fn:Eugeniy Khvastunov
n:Khvastunov;Eugeniy
org:Digma;IT
adr:;;;Kharkov;Kh;;Ukraine
email;internet:[EMAIL PROTECTED]
title:System Administrator
tel;work:+380675745646
tel;cell:+380504063116
version:2.1
end:vcard

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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Giorgio Incantalupo]

Problema solved!

Just put resetinterval=never inside zapata.conf


Giorgio Incantalupo


Giorgio Incantalupo wrote:

Hi,
I installed an Asterisk box with a sangoma A102 PRI card. Sometimes 
Asterisk drops calls...there is nothing inside logs but these warnings:


Sep 11 15:00:18 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.
Sep 11 15:00:22 WARNING[3503] chan_zap.c: Got restart ack on channel 
0/6 span 1 with owner
Sep 11 15:00:30 WARNING[3503] chan_zap.c: Ring requested on channel 
0/3 already in use on span 1.  Hanging up owner.
Sep 11 15:29:38 WARNING[3497] chan_sip.c: Maximum retries exceeded on 
transmission [EMAIL PROTECTED]

.168.3.175 for seqno 2 (Critical Response)
Sep 11 15:30:04 WARNING[3497] chan_sip.c: Maximum retries exceeded on 
transmission [EMAIL PROTECTED]

2.168.3.175 for seqno 2 (Critical Response)
Sep 11 15:30:24 WARNING[3497] chan_sip.c: Maximum retries exceeded on 
transmission [EMAIL PROTECTED]

2.168.3.175 for seqno 2 (Critical Response)
Sep 11 15:31:34 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.
Sep 11 16:00:08 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.
Sep 11 16:00:26 WARNING[3503] chan_zap.c: Got restart ack on channel 
0/6 span 1 with owner
Sep 11 16:03:13 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.

Sep 11 16:03:15 WARNING[15530] app_dial.c: Unable to forward voice
Sep 11 16:07:09 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.

Sep 11 16:13:22 WARNING[15925] app_dial.c: Unable to forward voice
Sep 11 16:15:37 WARNING[3503] chan_zap.c: Ring requested on channel 
0/6 already in use on span 1.  Hanging up owner.

Sep 11 16:21:26 WARNING[3504] chan_zap.c: Call specified, but not found?
Sep 11 16:23:21 WARNING[16267] app_dial.c: Unable to forward

Anyone ever got these messages? What do they mean? How can I fix them?

TIA

Giorgio Incantalupo

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[Asterisk-Users] Junghanns BRI cards and misdn

[Olivier]

Hi,Who has experienced using misdn instead of bristuff with Junghanns BRI cards inside a 1.2 Asterisk server ?What was it like ?Any advice about that ?Regards
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Re: [asterisk-users] DID not getting passed?

[Bob Chiodini]

On Mon, 2006-09-11 at 20:25 -0700, Christopher Corn wrote:
 im having issues when routing calls from the outside with my new VSP.
 this is what asterisk tells me when i try to make an incoming call, i
 get the no service response when i  call.
 
 -- Executing GotoIf(SIP/christopher_corn-eddb, 1?from-trunk||
 1) in new stack
 -- Goto (from-trunk,s,1)
 -- Executing NoOp(SIP/christopher_corn-eddb, No DID or CID
 Match) in new stack
 -- Executing Answer(SIP/christopher_corn-eddb, ) in new stack
 -- Executing Wait(SIP/christopher_corn-eddb, 2) in new stack
 -- Executing Playback(SIP/christopher_corn-eddb, ss-noservice)
 in new stack
 -- Playing 'ss-noservice' (language 'en')
 
  
 my extensions additional.conf has this
 [ext-did]
 include = ext-did-custom
 exten = 408335,1,Set(FROM_DID=4083354290)
 exten = 408335,n,Goto(ext-local,103,1)
 exten = s,1,Noop(No DID or CID Match)
 exten = s,n,Answer
 exten = s,n,Wait(2)
 exten = s,n,Playback(ss-noservice)
 exten = s,n,SayAlpha(${FROM_DID})
 exten = _[*#X].,1,Set(FROM_DID=${EXTEN})
 exten = _[*#X].,n,Noop(Received an unknown call with DID set to
 ${EXTEN})
 exten = _[*#X].,n,Goto(ext-did,s,1)
 ; end of [ext-did]
  
  
 i tried to replacing my number with my username, my phone number
 without area code, using dashes, but nothing works. 
  
 is it because my vsp, axvoice.com doesn't pass did's?
  
 any information is appreciated. thanks.
 

Try turning on SIP debug to see what you are getting from your provider.

Bob...
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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Steve Davies]

For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.

Thanks,
Steve

On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Problema solved!

Just put resetinterval=never inside zapata.conf


Giorgio Incantalupo


Giorgio Incantalupo wrote:
 Hi,
 I installed an Asterisk box with a sangoma A102 PRI card. Sometimes
 Asterisk drops calls...there is nothing inside logs but these warnings:

 Sep 11 15:00:18 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 15:00:22 WARNING[3503] chan_zap.c: Got restart ack on channel
 0/6 span 1 with owner
 Sep 11 15:00:30 WARNING[3503] chan_zap.c: Ring requested on channel
 0/3 already in use on span 1.  Hanging up owner.
 Sep 11 15:29:38 WARNING[3497] chan_sip.c: Maximum retries exceeded on
 transmission [EMAIL PROTECTED]
 .168.3.175 for seqno 2 (Critical Response)
 Sep 11 15:30:04 WARNING[3497] chan_sip.c: Maximum retries exceeded on
 transmission [EMAIL PROTECTED]
 2.168.3.175 for seqno 2 (Critical Response)
 Sep 11 15:30:24 WARNING[3497] chan_sip.c: Maximum retries exceeded on
 transmission [EMAIL PROTECTED]
 2.168.3.175 for seqno 2 (Critical Response)
 Sep 11 15:31:34 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 16:00:08 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 16:00:26 WARNING[3503] chan_zap.c: Got restart ack on channel
 0/6 span 1 with owner
 Sep 11 16:03:13 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 16:03:15 WARNING[15530] app_dial.c: Unable to forward voice
 Sep 11 16:07:09 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 16:13:22 WARNING[15925] app_dial.c: Unable to forward voice
 Sep 11 16:15:37 WARNING[3503] chan_zap.c: Ring requested on channel
 0/6 already in use on span 1.  Hanging up owner.
 Sep 11 16:21:26 WARNING[3504] chan_zap.c: Call specified, but not found?
 Sep 11 16:23:21 WARNING[16267] app_dial.c: Unable to forward

 Anyone ever got these messages? What do they mean? How can I fix them?


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Re: [asterisk-users] Whcih phones are better for mass deployment

[picciuX]

well, not to disappoint anyone, but GrandStream Phones DO support remote provisioning. It's only a matter of setting it up. But with tftptext editor you can set every feature and update firmware remotely.You could also reboot the phone using CURL.
Anyway, i think in new GXP-2000 firmwares also reboot via SIP-NOTIFY is supported.2006/9/10, Thomas Kenyon [EMAIL PROTECTED]
:Alberto Sagredo wrote: I prefer Linksys ones. Spa 9xx series, are great, and provisioning from
 Sipura/Linksys is much better than PA1628 (Unencrypted). Supports https,tftp and http. With Encryption. Vonage use it.I'm sure they are much better, they should be they cost a lot more. I
was merely expressing surprise that the Grandstream phones couldn't.Though out of cursiousity, in the context of this thread, What advantagewould being able to encrypt the data provide for the person asking the
original question?Surely that is only of benefit if you are an ITSP, of no use at all foran office installation.___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] Verify Database Installation

[broadbandvoice]

This is a questions about database verification and not a2billing. Asterisk also uses database for such things as cdr and sometimes you call dial plans from database. Someone might have seen a similar situation while installing postgres for Asterisk. It is Asterisk related. 

-- Original message -- From: "Areski K" [EMAIL PROTECTED]  Please try to redirect those questions to the appropriate place,  I mean the A2Billing forum : http://forum.asterisk2billing.org  It's off-topics for the Asterisk-user mailing-list.   Kind regards,  /AreskiOn 9/11/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>wrote:   Everything was going well, I got the tarball, unpacked the tarballs, created   the postgre user and password, database is created and checked ownership and   even got a list of database users. I even imported the data schema into the   new database. My problem now is verification of database installation. I get   an error below when i try it:  
 t;   a2billing= SELECT * FROM cc_ui_authen;   ERROR: relation "cc_ui_authen" does not exist   a2billing=   -- Original message --   From: [EMAIL PROTECTED] You're right. How did I miss that? -- Original message --   From: [EMAIL PROTECTED] Yes, I put the filename   "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got   that error. -- Original message --   From: "Jamin W. Collins" <[EMAIL PROTECTED]> [EMAIL PROTECTED] wrote: I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all  &
 gt; 
t;  total 4872 drwxr-xr-x 2 root root 4096 Sep 11 06:22 . drwxr-xr-x 20 root root 4096 Sep 10 21:28 .. -rw-r--r-- 1 root root 165 Sep 11 06:16   download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz -rw-r--r-- 1 root root 4960345 Sep 11 06:31   download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz  ^The above is your file name, note the additional "download.php?get=" onthe file name.   --   a mp;a m p;g t ; Jam in W. Collins___--Bandwidth and Colocation provided by Easynews.com --  
  aster
isk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users -- Forwarded message --   From: [EMAIL PROTECTED]   To: Asterisk Users Mailing List - Non-Commercial Discussion     Date: Mon, 11 Sep 2006 14:23:43 +   Subject: Re: [asterisk-users] Problems Unpacking tarball For Asterisk   Application   ___   --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 -- Forwarded message -- 
 
t;  From: [EMAIL PROTECTED]   To: Asterisk Users Mailing List - Non-Commercial Discussion     Date: Mon, 11 Sep 2006 14:17:25 +   Subject: Re: [asterisk-users] Problems Unpacking tarball For Asterisk   Application   ___   --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 -- asterisk-users mailing list   To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/ast
 erisk-
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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Rich Adamson]

Steve Davies wrote:

For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.

Thanks,
Steve

On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Problema solved!

Just put resetinterval=never inside zapata.conf


Giorgio Incantalupo


If memory serves correctly, I believe the parameter was added a couple 
of years ago as a means / workaround for hung channels. At the time, 
there was not any overwhelming evidence as why a channel would 
occasionally hang. Some of the possibilities included unusual 
interaction from the opposite end of the T1/E1, anomalies in the 
dialplan, etc.


Now that a substantial amount of work / changes have been made relative 
to PRI's and other internal asterisk code, there appears to be less of a 
need to reset.


A reasonable approach might be to apply the parameter and pay close 
attention to channels that might be in some strange state. If none are 
observed, then leave it.


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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Steve Davies]

On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:

Steve Davies wrote:
 For the curious, can anyone tell me how this flag fixes the issue? - I
 have seen the error before, but always assumed it was related to hung
 channels.

 Thanks,
 Steve

 On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
 Problema solved!

 Just put resetinterval=never inside zapata.conf


 Giorgio Incantalupo

If memory serves correctly, I believe the parameter was added a couple
of years ago as a means / workaround for hung channels. At the time,
there was not any overwhelming evidence as why a channel would
occasionally hang. Some of the possibilities included unusual
interaction from the opposite end of the T1/E1, anomalies in the
dialplan, etc.

Now that a substantial amount of work / changes have been made relative
to PRI's and other internal asterisk code, there appears to be less of a
need to reset.

A reasonable approach might be to apply the parameter and pay close
attention to channels that might be in some strange state. If none are
observed, then leave it.


Thanks for that. I have a customer who is using Asterisk 1.0.x, and I
am tempted to backport this fix from the 1.2.x code where it was
introduced.

Cheers,
Steve
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Re: [asterisk-users] Grandstream Budgetone phones don't show

[Ricardo Carvalho]

Thanks Jessee,

I've just sent an e-mail to Grandstream support asking if they are 
planning in a near future to release a firmware implementing 
alphanumeric callerid for Budgetone series.
When they answer me, I'll replay to this thread with their feedback, so 
the community can also benefit...


Regards,
Ricardo.






Jessee J Holmes wrote:

Ricardo,

From what I know its a physical limitation of the display Grandstream 
chose on that phone, Grandstream recommends purchasing the GXP-2000 
phone instead if you're looking for this feature.


Grandstream has no plans from what I am aware of of making this change 
to the BudgetTone series phones.


You are more than welcome to inquire directly from Grandstream though, 
this is just from what I know from dealing with them in the past.



Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


Looking for voice over IP products?  Visit our VoIP store at 
http://voipstore.atacomm.com/




On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote:

I guess this functionality will be in the future added to new 
firmware releases don't you people think so?


Ricardo.






Doug Lytle wrote:


These phones aren't capable of alphanumeric entries, only numeric.

Doug


Tom Vile wrote:

They only do numeric callerid.





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[Asterisk-Users] Which SIP hardphone implements RTCP XR (aka RFC3611)

[Olivier]

Hi,RFC3611 provides a way to monitor call quality.Do you know any SIP hardphone implementing this feature ?I'm aware of softphones doing so (Counterpath's EyeBeam, for example) but no hardphone yet.
Cheers
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Re: [Asterisk-Users] Junghanns BRI cards and misdn

[Giorgio Incantalupo]

Hi Olivier,
I used bristuff then I passed to misdn.
But there are pros and cons:
- misdn is easier to install and configure (I do not know if bristuff 
installation has been improved...but it was a bit tricky when I used it..)
- misdn has its own config file (misdn.conf) and does not use zap 
channels (if you do not use extensions.conf is a bit more of work!)



Consider that I have always used OEM monoBRI cards or beronet cards, 
even with bristuff  (it seems that chipset is similar)...and after 
installing I have not had problems that beronet support team could not 
solve.



Giorgio Incantalupo




Olivier wrote:

Hi,

Who has experienced using misdn instead of bristuff with Junghanns BRI 
cards inside a 1.2 Asterisk server ?

What was it like ?
Any advice about that ?

Regards


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Re: [asterisk-users] How to notify an ACD agent before he/she picks up

[picciuX]

doing it the dialplan way:if you log in your agents via AgentCallBackLogin, you can set the CallBack extension to an extension managed by a macro, where the macro will do what you need:extensions.conf
[agents-exts]exten = 100,1,Macro(stdagent|SIP/100|...)exten = 101,1,Macro(stdagent|SIP/101|...)[macro-stdagent]exten = s,1,AGI(notify-agent)exten = s,2,Dial(${ARG1}|...)
[agents-login]exten = 500,1,AgentCallBackLogin(||${CALLERID(number)[EMAIL PROTECTED])hope this helps...2006/9/12, Watkins, Bradley 
[EMAIL PROTECTED]:In the forthcoming 1.4, you can tell the Queue application to run an AGI just before sending the call to the destination.In the AGI, you can use the (also new in 
1.4) MEMBERINTERFACE channel variable to determine the destination.Of course, that's not a solution now since 1.4 is not even beta yet.But I figured I'd present another possibility.- Brad
From: [EMAIL PROTECTED] on behalf of Richard LymanSent: Mon 9/11/2006 6:37 PMTo: 
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] How to notify an ACD agent before he/she picks upMF wrote: Has anyone got a clue about this?I need to know which operator to
 send a message to,prior to the queue command ringing him,(just after he is assigned) Anyone knows if I can get to know the operator ACD choosed to send the call by usingRealtime Queue, or maybe via the manager?
someone already told you to look at the manager.make sure queues.conf has eventwhencalled=yesthen you will get a manager eventEvent: AgentCalled...___
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[asterisk-users] Rack for Asterisk with TDM2400 Digium board

[Antoine Megalla]

Hi,

I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so 
can anyone suggest a barebone 1U or 2U server (I
prefer the SuperMicro 
Superservere series) with a Mother board that is
compatible with the Digium 
TDM2400 card (which is a Full Length PCI card).

Thank you and best regards,

Antoine Megalla 



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Re: [asterisk-users] Rack for Asterisk with TDM2400 Digium board

[BJ Weschke]

On 9/12/06, Antoine Megalla [EMAIL PROTECTED] wrote:

Hi,

I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a barebone 1U or 2U server (I
prefer the SuperMicro
Superservere series) with a Mother board that is
compatible with the Digium
TDM2400 card (which is a Full Length PCI card).



I wouldn't recommend anything less than a 4RU machine from SuperMicro
with the size of the TDM2400 card.


--
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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Rich Adamson]

Steve Davies wrote:

On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:

Steve Davies wrote:
 For the curious, can anyone tell me how this flag fixes the issue? - I
 have seen the error before, but always assumed it was related to hung
 channels.

 Thanks,
 Steve

 On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
 Problema solved!

 Just put resetinterval=never inside zapata.conf


 Giorgio Incantalupo

If memory serves correctly, I believe the parameter was added a couple
of years ago as a means / workaround for hung channels. At the time,
there was not any overwhelming evidence as why a channel would
occasionally hang. Some of the possibilities included unusual
interaction from the opposite end of the T1/E1, anomalies in the
dialplan, etc.

Now that a substantial amount of work / changes have been made relative
to PRI's and other internal asterisk code, there appears to be less of a
need to reset.

A reasonable approach might be to apply the parameter and pay close
attention to channels that might be in some strange state. If none are
observed, then leave it.


Thanks for that. I have a customer who is using Asterisk 1.0.x, and I
am tempted to backport this fix from the 1.2.x code where it was
introduced.


From a personal perspective, I think I'd hold off on the back port and 
devote that time towards testing the soon to be released version (now in 
Trunk).


If you've watched the number and type of changes that have gone into SVN 
Trunk in the last couple of months, it appears as though a significant 
number of possible memory leaks, sip code, infrastructure code, PRI code 
changes, etc, have been applied that would be beneficial for all 
production systems. There also appears to be a fair amount of work that 
will be needed to upgrade dialplan syntax (etc) for the new release.


Best guess is that once the Trunk code gets past the beta testing phase, 
it will likely be the asterisk code of choice for most/all production 
systems.


Consider the above is only my $0.02 worth. ;)

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Re: [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)

[John Marvin]

shadowym wrote:


I found that the distortion was consistent.  In other words it happened in
the same way at the same time in a particular file.  I suspect it has
something to do with how Asterisk plays it back and not any sort of
hardware/IDE/interrupt issue.  Kris, the developer of Astlinux didn't seem
to have any ideas why it would not work as well on Asterisk either.


If the distortion is consistant as you say, then you are probably seeing 
the same problem I found a workaround for.


I was disappointed in the sound quality of the gsm files, so I was happy 
to find the Native Sounds files. However, I then ran into the clicks and 
pops when playing them. Reading some of the earlier comments in this 
discussion someone mentioned an issue with Asterisk not padding files to 
even 20ms increments when playing them. So, although that may be a bug 
in Asterisk, I thought I would see if that was the problem by writing a 
quick C program to pad all my ulaw files to multiples of 160 bytes. 
Voila, all clicks and pops were gone. So, I don't know if that is the 
only issue, and perhaps there are other problems people are having, but 
padding the files fixed the issue for me. Obviously this should be fixed 
in Asterisk.


If anyone else wants to try this experiment I've enclosed the simple C 
program I wrote below. If you compile it and call it padulaw here is how 
I fixed all the files:


find /var/lib/asterisk/sounds -type f -name '*.ulaw' | xargs padulaw

This program could be easily modified to pad .sln files to a multiple of 
320 bytes (the files would be padded with 0x rather than 0xff).


John

#include stdio.h
#include fcntl.h
#include sys/stat.h
#include sys/types.h

#define ULAW_SILENCE 0xff
#define MS20_BYTES  160

unsigned char silence[MS20_BYTES];

void pad_file(char *);

main(int argc, char **argv)
{
int i;
int nfiles;

if (argc  2) {
fprintf(stderr,Usage: %s file name ...\n,argv[0]);
exit(1);
}

nfiles = argc - 1;
for (i = 0; i  nfiles; i++) {
pad_file(argv[i+1]);
}

exit(0);
}

void
pad_file(char *fname)
{
int fd;
int i;
struct stat sbuf;
int filesize;
int remainder;
int nwrite;

fd = open(fname,O_WRONLY|O_APPEND);
if (fd  0) {
fprintf(stderr,Could not open %s for writing.\n,fname);
return;
}

if (fstat(fd,sbuf) != 0) {
fprintf(stderr,Could not stat file %s.\n,fname);
return;
}

filesize = (int) sbuf.st_size;
remainder = filesize % MS20_BYTES;
if (remainder == 0) {
close(fd);
return;
}

nwrite = MS20_BYTES - remainder;
for (i = 0; i  nwrite; i++)
silence[i] = (unsigned char)ULAW_SILENCE;

if (write(fd,(void *)silence,nwrite) != nwrite) {
fprintf(stderr,Write Failure on file %s\n,fname);
return;
}

close(fd);
return;
}
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Re: [asterisk-users] Static RealTime - SIP.CONF

[Hugo]

BenjaminI've
already read all voip-info's articles. The address you've mentioned
shows how to configure a DYNAMIC RealTime, not a STATIC one.

I've tried to use the same table with both realtime modules, but it didn't work. No users have been found (sip show conf).

If you could help me to solve my problem, I would be tkankful
regards

2006/9/12, Benjamin Jacob [EMAIL PROTECTED]:
Rushowr wrote:Hugo wrote:Anyone could help to use Static RealTime with SIP.CONF. I use DynamicRealtime successfully. In fact, I want to know how to compos the correct
DB(postgres or mysql) fields (I think STATIC configuration is differentfrom DYNAMIC).Regards,Hugo
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www.voip-info.org is an amazing tool, and it's referenced FREQUENTLY.10 seconds in a web browser brought me this link:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipWhich, amazingly enough, contains information about setting up thetables for RealTime Sip
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip talks abt Realtime (dynamic) config.For static config :http://www.voip-info.org/wiki-Asterisk+RealTime
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Re: [asterisk-users] Polycom Soundpoint Key Remap

[Adam Goryachev]

Shawn Kelley wrote:


Hi,

Does anyone know how to do a re-map of a key on the Polycom to make it 
dial a number.


I know how to remap a key to a certain function, but I don’t know how 
to make it dial a number.


I’m wanting to re-map the “Service” key to dial *8 for a group pickup.

Any help is greatly appreciated.

Thanks!

--Shawn

AFAIK, you will need to tell it save a speed dial for *8, and then map 
the key to dial the speed dial number that you saved it as.


Hope this helps.

Regards,
Adam

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Re: [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)

[Steve Davies]

On 9/12/06, John Marvin [EMAIL PROTECTED] wrote:

shadowym wrote:

[snip]

Asterisk not padding files to
even 20ms increments when playing them. So, although that may be a bug
in Asterisk, I thought I would see if that was the problem by writing a
quick C program to pad all my ulaw files to multiples of 160 bytes.
Voila, all clicks and pops were gone. So, I don't know if that is the
only issue, and perhaps there are other problems people are having, but
padding the files fixed the issue for me. Obviously this should be fixed
in Asterisk.

If anyone else wants to try this experiment I've enclosed the simple C
program I wrote below. If you compile it and call it padulaw here is how
I fixed all the files:

find /var/lib/asterisk/sounds -type f -name '*.ulaw' | xargs padulaw

This program could be easily modified to pad .sln files to a multiple of
320 bytes (the files would be padded with 0x rather than 0xff).

John


[snip]

I don't suppose you know what the silence padding bytes would be for ALAW?

Thanks,
Steve
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[asterisk-users] Features.. phone vs. asterisk?

[Nick Ellson]

I tried a lot of SIP and IAX softphones looking for ones I liked, noticing 
some have certain features and others did not. For things like call 
transfer, call park, group pick-up, line presence, and all those kinds of 
extras I have a bit of confusion on where it is implemented?


Are these functions that Asterisk handles and the phone just triggers 
them with some out-of-band signal or DTMF sequence? Or does some of this 
rest on the phone itself? (Here is where I would love TFM to R. :) Just 
having a hard time finding what to read.)




Nick


--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.

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[asterisk-users] WG: Asterisk and Agents

[mbodbg]

Hello NG,

We've a small problem using agents in asterisk. One requirement is, if there
no agent logged into a queue, it shouldn't be possible that a call joins a
queue. I can configure that using the parameter joinempty=strict in
queues.conf, unfortunately the parameter takes only effect if I add
members to the queue dynamically. If there are static members assigned to
the queue, a call can always join the queue, even if there are no agents
logged. To add agents dynamically to a queue I'm using the following scripts
in the dialplan:

exten = _*8XXX,1,Answer
exten = _*8XXX,2,SetLanguage(de)
exten = _*8XXX,3,AddQueueMember(DEMO|Agent/${EXTEN:1})
exten = _*8XXX,4,Dial(Local/999/n,,D(#))
exten = _*8XXX,5,AgentCallBackLogin(${EXTEN:1}|[EMAIL PROTECTED]) 
exten = _*8XXX,6,Hangup()

exten = _**8XXX,1,Answer
exten = _**8XXX,2,SetLanguage(de)
exten = _**8XXX,3,RemoveQueueMember(DEMO|Agent/${EXTEN:2})
exten = _**8XXX,4,AgentCallbackLogin(${EXTEN:2})
exten = _**8XXX,5,Hangup()

So if I type e.g. *8000 it logs in Agent/8000 and adds the agent dynamically
to the queue test. With **8000 it logs out Agent/8000 and removes the agent
from queue test. All that work's fine. The problem is that to add an agent
to a queue, it has NOT to be defined in agents.conf. If an agent mistypes
his agent ID, e.g *8999, it logs on Agent/8999, even if it is not defined in
agents.conf. As result Agent/8999 keeps assigned to the queue DEMO, and
because there is an agent assigned the parameter joinempty has no effect
anymore. Calls can join the queue even if there is no real Agent logged in.
Any ideas are welcome.

Best Regards

- Markus   



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Re: [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.

[Steve Davies]

On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Hi,
I get many of these warnings inside Asterisk log:
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
0/1 already in use on span 1.  Hanging up owner.


What does they mean??



Can I assume then that 'resetinterval=never' did not make this problem go away?

Cheers,
Steve
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[asterisk-users] How to setup announce attibute in queues.conf

[gc]

I have this line in my queues.conf:
announce= support-department
and I have an recording file 
support-department-recording.wav file.
Can anybody tell me how to setup support-department 
so it play the .wav file when agent pickup the phone? Where should I define 
support-department so asterisk will play support-department-recording.wav? Is 
this in musiconhold.conf?

gc

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Re: [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)

[Steve Davies]

On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:


I don't suppose you know what the silence padding bytes would be for ALAW?


Found it... It is 0x55.

Thanks for the program :)
Steve
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Re: [asterisk-users] Dell hardware ...

[Brodie Macleod]

I'm using a Dell SC1430 that includes the Intel NIC and don't have any 
problems at all. Also using a TE210P and TDM400P w/ 4 FXS in the box.  I've 
never had to reboot the box or restart Asterisk (except for kernel upgrades 
and * upgrades of course).

-Brodie


On Monday 11 September 2006 05:12 pm, Alan Bunch wrote:
 I was going to use a Dell 1425 for Asterisk build but I see on Digium's
 website that hardware may be problematic.  Can anyone shed a litle more
 light on the problem.   I see the Intel ethernet cards seem to cause
 problems.  If I need to disable the onboard Intel on the Dell hardware I
 can I just need to know what to expect.

 How about the 850, any word there  ?

 TIA

 Alan




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RE: [asterisk-users] Dell hardware ...

[Arjan Kroon]

Hi, Alan,

We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
in it and it works perfect.
It is almost PlugPlay.

greetings


Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: dinsdag 12 september 2006 8:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dell hardware ...

Hi Alan,
simply do not use Dell hardware.
we had your problem, we called Dell and they told us that our servers 
was not configurable (they were too cheap).
So now I do not use Dell anymore and we have less problem.


Giorgio Incantalupo


Alan Bunch wrote:
 I was going to use a Dell 1425 for Asterisk build but I see on 
 Digium's website that hardware may be problematic.  Can anyone shed a 
 litle more light on the problem.   I see the Intel ethernet cards seem

 to cause problems.  If I need to disable the onboard Intel on the Dell

 hardware I can I just need to know what to expect.

 How about the 850, any word there  ?

 TIA

 Alan




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[asterisk-users] Deploying an IVR - direct extensions.conf or AGI scripts?

[Marco Mouta]

Hi all,I'm developing an IVR that will have to make some MYSQL queries and diferent DTMF menus. Preventing already my development effort, future I plan to deploy my own website where users can build their own IVR.
Would you recomend me to make it with Realtime Extensions, do it directly in extensions.conf and for queries and something else use AGI scripts, or you recomend me to build specific AGIscripts with IVR menus inside (this looks very limited for future WebConfig interface)?
What is your advice, concerning with your experience.-- Best regards,Marco Mouta
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RE: [BULK] Re: [asterisk-users] Prompts recording for Asterisk

[Savoy, Kevin - Williston, ND]

Is there a way to contact her directly or do we have to go through
Digiums website?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Sunday, August 27, 2006 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [BULK] Re: [asterisk-users] Prompts recording for Asterisk
Importance: Low

snip
 2) What are the best sources (cost effective) to get prompts recorded.
/snip
I would go with allison. She is the one that did all the voice files
that 
you currently have on asterisk. So if you use her for your prompts you
will 
have the same voice thru out ur PBX. A client of mine just used her for
his 
entire pbx (total of 12 clips i believe ranging in sizes). The price was

$75.00 

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RE: [asterisk-users] Polycom Soundpoint Key Remap

[Douglas Garstang]

 -Original Message-
 From: Adam Goryachev [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, September 12, 2006 6:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap
 
 
 Shawn Kelley wrote:
 
  Hi,
 
  Does anyone know how to do a re-map of a key on the Polycom 
 to make it 
  dial a number.
 
  I know how to remap a key to a certain function, but I 
 don’t know how 
  to make it dial a number.
 
  I’m wanting to re-map the “Service” key to dial *8 for a 
 group pickup.
 
  Any help is greatly appreciated.
 
  Thanks!
 
  --Shawn
 
 AFAIK, you will need to tell it save a speed dial for *8, and 
 then map 
 the key to dial the speed dial number that you saved it as.

Except... if you want to send DTMF digits during a call. We wanted to map the 
transfer key on the Polycom to send #2 for an Asterisk assisted tranfers, as 
transfering in Queues is known to completey destroy Asterisk Queues until a 
restart. Programming the Transfer to to send a speed dial (of #2) would 
generate a new call to an extension, #2.

Doug.
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RE: [asterisk-users] Grandstream Budgetone phones don't show

[Jessee Holmes]

Great! Much appreciated, I'll do some investigation myself, I'll be visiting 
Grandstream this week.

Jessee J Holmes

-Original Message-
From: Ricardo Carvalho [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: 9/12/06 7:11 AM
Subject: Re: [asterisk-users] Grandstream Budgetone phones don't show

Thanks Jessee,

I've just sent an e-mail to Grandstream support asking if they are 
planning in a near future to release a firmware implementing 
alphanumeric callerid for Budgetone series.
When they answer me, I'll replay to this thread with their feedback, so 
the community can also benefit...

Regards,
Ricardo.






Jessee J Holmes wrote:
 Ricardo,

 From what I know its a physical limitation of the display Grandstream 
 chose on that phone, Grandstream recommends purchasing the GXP-2000 
 phone instead if you're looking for this feature.

 Grandstream has no plans from what I am aware of of making this change 
 to the BudgetTone series phones.


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[Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk

[Olivier]

Hi,What would you suggest to implement directed call pickup on bristuffed Asterisk 1.2 ?I'm after tle ability to pick a specific ringing call (without caring about which call arrived first, for example).
Something like : *8 + local extension would be perfect.voip-info.org introduces many paths (http://bugs.digium.com/view.php?id=5014
, http://linux.thorsten-knabe.de/asterisk/pickup.jsp, ...) but which should be more stable ?Cheers
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RE: [BULK] Re: [asterisk-users] Prompts recording for Asterisk

[Dean Collins]

Email her directly [EMAIL PROTECTED] 

Don't forget to 'donate' the recordings back to Digium for inclusion.

 

Cheers,

Dean

 


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND
 Sent: Tuesday, 12 September 2006 10:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [BULK] Re: [asterisk-users] Prompts recording for
Asterisk
 
 Is there a way to contact her directly or do we have to go through
 Digiums website?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dovid
 Bender
 Sent: Sunday, August 27, 2006 11:30 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [BULK] Re: [asterisk-users] Prompts recording for Asterisk
 Importance: Low
 
 snip
  2) What are the best sources (cost effective) to get prompts
recorded.
 /snip
 I would go with allison. She is the one that did all the voice files
 that
 you currently have on asterisk. So if you use her for your prompts you
 will
 have the same voice thru out ur PBX. A client of mine just used her
for
 his
 entire pbx (total of 12 clips i believe ranging in sizes). The price
was
 
 $75.00
 
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Re: [asterisk-users] More Zaptel build problems

[Kristian Kielhofner]

Kristian Kielhofner wrote:

Hello everyone,

I am trying to build zaptel 1.2.9 for AstLinux.  I have already done 
an svn export of the 1.2.9 tag, so I am not experiencing the missing 
octastic issue.


However, I am having a funny problem.  The zaptel.log that I have 
attached tells the full story, but I'll give you a synopsis...


- Because I need to cross compile and the Zaptel Makefile does not 
really have a concept of CC/HOSTCC, I have to build the makefw 
gendigits tor2fw.h radfw.h targets with my HOSTCC - gcc.  I then build 
the rest of zaptel using the normal uclibc cross compiler.  This has 
always worked until now.


- AstLinux doesn't have hotplug, so I have to define 
HOTPLUG_FIRMWARE=no.  This means that vpm450m.c has to include 
vpm450m_fw.h which, as shown in the compiler output attached, has some 
syntax errors:


/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m_fw.h:1: 
error: syntax error before '/' token
In file included from 
/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m.c:16: 

/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m_fw.h:1:75: 
too many decimal points in number
/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m.c: 
In function `init_vpm450m':
/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m.c:405: 
error: `vpm450m_fw' undeclared (first use in this function)
/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m.c:405: 
error: (Each undeclared identifier is reported only once
/home/kris/projects/astlinux-trunk/build_i586/zaptel-1.2.9/wct4xxp/vpm450m.c:405: 
error: for each function it appears in.)



My guess is that my hack of building makefw and friends might not 
work anymore...  Or maybe this is some kind of strange bug.  Any ideas?


Thanks!

--
Kristian Kielhofner



Kristian (replying to my own post),

	It seems that this error was from fw2h.c putting the full pathname in 
the generated vpm450m_fw.h.  Although I just spent 20 minutes making my 
own patch, it seems that it was fixed a while ago in r1458.  Woo hoo 
zaptel 1.2.9.1!


--
Kristian Kielhofner
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[asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains

[harrygaillac-sip]

I've ever post this question many times on asterisk
users without success ?

My config :

SER = outbound proxy presence/im server

   ASTERISK
|| 
||   
  proxy/SER ===sip agents
  +
   rtpproxy

If a sip agents dial local uri no problems but if
those sip agents want to dial a none local uri ser
have to handle the requests.

I want to all sip requests for any domains are sent to
asterisk if domain is non local  asterisk forward the
request to ser in oder to handle the transaction
between callee==ser==asterisk==caller


[sip]

exten = _.,1,NoOp(Incoming Call from house extension
${CALLERID} for [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${SIPDOMAIN} = nxs.yi.org]?3:4)
exten = _.,3,Goto(sip-local,${EXTEN},1)
exten = _.,4 Goto(outbound)
exten = h,1,HangUp()
.
 
I have to set up a context with outboundproxy !
any idea to write it ?

harry



  I use asterisk svn-trunk .
 
  I wish asterisk to forward all sip requests from
 non
  local domains to a proxy .
 
  For example asterisk handle domainA a sip agent
 send a
  invite to a domainB .
 
  Is asterisk able to check the domain and so
 forward
  the request to a context (with outboubounproxy)
 
 
 This is not a -dev question. We never ever forward
 SIP requests, you  
 need
   a SIP proxy for that. You are well aware of that
 fact, Harry.
 
 /Olle
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Re: [asterisk-users] Grandstream Budgetone phones don't show

[Craig Guy]

The lcd in the current budgetone series cannot support alphnumeric display.

Craig

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

Sent: Tuesday, September 12, 2006 8:11 PM
Subject: Re: [asterisk-users] Grandstream Budgetone phones don't show



Thanks Jessee,

I've just sent an e-mail to Grandstream support asking if they are 
planning in a near future to release a firmware implementing alphanumeric 
callerid for Budgetone series.
When they answer me, I'll replay to this thread with their feedback, so 
the community can also benefit...


Regards,
Ricardo.






Jessee J Holmes wrote:

Ricardo,

From what I know its a physical limitation of the display Grandstream 
chose on that phone, Grandstream recommends purchasing the GXP-2000 phone 
instead if you're looking for this feature.


Grandstream has no plans from what I am aware of of making this change to 
the BudgetTone series phones.


You are more than welcome to inquire directly from Grandstream though, 
this is just from what I know from dealing with them in the past.



Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


Looking for voice over IP products?  Visit our VoIP store at 
http://voipstore.atacomm.com/




On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote:

I guess this functionality will be in the future added to new firmware 
releases don't you people think so?


Ricardo.






Doug Lytle wrote:


These phones aren't capable of alphanumeric entries, only numeric.

Doug


Tom Vile wrote:

They only do numeric callerid.





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[asterisk-users] Re: RE : Re: [asterisk-dev] Forwarding sip requests from none local domains

[Tzafrir Cohen]

On Tue, Sep 12, 2006 at 04:33:21PM +0200, [EMAIL PROTECTED] wrote:
 I've ever post this question many times on asterisk
 users without success ?

asterisk-dev is not 2-level support for asterisk-users . Please
follow-up there.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] IAX phone recommandation

[richard Coco]

Hi all,

we plan to install several IAX softphones.

http://www.voip-info.org/wiki-Asterisk+IAX+clients
lists a lot of IAX phones for Windows and Linux. Which
one would you recommand? We will install IAX client on
Linux and Windows.

thx richard

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[asterisk-users] Dropped call question - Maximum retries exceeded on transmission

[Kohler, Jeffrey]

I am encountering an intermittent issue where some of my calls are being
dropped.  Most of the calls that are made are successful.  However, some
calls will be dropped after having been connected for some time.

Each time a call gets dropped, I get output similar to the following in
the Asterisk console:

Sep 12 18:52:36 WARNING[4620]: chan_sip.c:1835 retrans_pkt: Maximum
retries exceeded on transmission  for seqno 1620 (Critical Response)

Sep 12 18:52:36 WARNING[4620]: chan_sip.c:1835 retrans_pkt: Hanging up
call  no reply to our critical packet.

Does anyone have any suggestions?  I honestly don't know where to start
investigating this issue, so if anyone has any ideas they would be
greatly appreciated.

Thanks
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[asterisk-users] Conference bridge problem

[Bartosz Wegrzyn - asterisk]

Hello,

I am trying to set conference system that will allow to bridge pstn and
voip conferences together,

So far I did this

created meetme conference room
conf = 500|1234


I created test extension 555, which does this:

exten = 555,1,MeetMeCount(500|count)
exten = 555,2,Gotoif,$[${count} = 0]?7
exten = 555,3,Gotoif,$[${count} = 1]?9
exten = 555,4,Meetme,500|cxAMs
exten = 555,5,Playback,goodbye
exten = 555,6,Hangup
exten = 555,7,Goto(from-internal-custom,556,1)
exten = 555,8,hangup
exten = 555,9,System(/usr/sbin/asterisk -rx meetme kick 500 2)
exten = 555,10,Goto(from-internal-custom,556,1)


1st check how many people are in meetme conference 500
if more than 1 skip to 9 if zero go to 7

this is done because if zap channel is still up (from previous conference)
and in the conference it will block new

conference connection to pstn.
so my way is to check if there is more than 1 user in the conference, if
yes it would mean
that zap channel is still up (this is my main problem , so thats why I do
that)

if it is up I will go to extension 9 and I will kill it before I proceed
later I will run this:

exten = 555,10,Goto(from-internal-custom,556,1)
which initiates zap call using this file:


[EMAIL PROTECTED] asterisk]# cat 1-test
Channel: ZAP/4/91(number deleted)
Callerid: 1
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: from-internal-custom
Extension: 561
Priority: 1


above would call pstn number and put the call into extension 561

extension 561 would later dial dtmf codes to connect to the conference
with password:

exten = 561,1,wait,10
exten = 561,2,senddtmf(2)
exten = 561,3,senddtmf(7)
exten = 561,4,senddtmf(2)
exten = 561,5,senddtmf(5)
exten = 561,6,senddtmf(7)
exten = 561,7,senddtmf(3)
exten = 561,8,senddtmf(6)
exten = 561,9,senddtmf(#)
exten = 561,10,Meetme,500|qAx|1234
exten = 561,11,Hangup


at 561,10 it would go back to conference

at this time user is connected to 555 conference which is bridged with
pstn conference


When new user connects he goes to extension
exten = 555,7,Goto(from-internal-custom,556,1)

which does:
exten = 556,1,System(/bin/cp /etc/asterisk/1-test
/var/spool/asterisk/outgoing/)
exten = 556,2,goto(from-internal-custom,555,4)



because there are more than 2 users in the 500 conference (1st user and
pstn user) more uses can connect to join the

bridge.

Problem!!!

When all users disconnect, the zap channel is still up.

it will be killed next time the new user connects to the conference.
During the silent time the zap channel will not be available.


So, I created temporary solution
i wrote this script:

[EMAIL PROTECTED] asterisk]# cat script
a=0
/usr/sbin/asterisk -rx meetme list 500  | grep Sip

if [ $? != 0 ];then
a=2
else
a=1
fi

/usr/sbin/asterisk -rx meetme list 500  | grep IAX

if [ $? != 0 ];then
a=2
else
a=1
fi

/usr/sbin/asterisk -rx meetme list 500  | grep Zap

if [ $? = 0 ];then
if [ $a = 2 ];then
/usr/sbin/asterisk -rx meetme kick 500 2
/usr/sbin/asterisk -rx meetme kick 500 1
/usr/sbin/asterisk -rx meetme kick 500 3
fi
fi


The script checks if zap channel is up, although the IAX or SIP are down.
If it is up it will kill the zap channel.

Problem is that running that script using cron starts a lot of rastersik
processes and
asterisk stop working,

Any ideas how my problem could be solved?

Thx

Bart






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Re: [Asterisk-Users] Junghanns BRI cards and misdn

[Olivier]

Thanks for your answer.Has anyone followed the other way (msidn
on Junghanns board), as this would certainly prevent Junghanns nor
anyone else to provide any kind of support.Cheers
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Re: [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.

[Steve Davies]

On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:

On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
 Hi,
 I get many of these warnings inside Asterisk log:
 WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
 0/1 already in use on span 1.  Hanging up owner.

 What does they mean??


Can I assume then that 'resetinterval=never' did not make this problem go away?


To expand on my interest this...

We have a number of nearly identical installations of Asterisk. Same
H/W, same Zaptel, and same Asterisk build etc etc - All of the PRI
hardware is the same (Sangoma A101U), and a huge percentage of E1
lines in the UK are terminated by British Telecom.

Even though there is this amount of comonality between them, we have
exactly one customer who gets the already in use message seen above
on a regular basis, and a second customer who had the error only once.
The error tends to be fatal for inbound calls as it leaves the
channel locked permanently and the telco continues to try to use it :(

In almost every case there is an obvious SIP conversation on the box
that has not cleared down fully, and which seems to be holding the Zap
channel open in error. In the first company, they use a lot of WiFi
phones, and in the 2nd company they used to use WiFi phones (different
model), but don't anymore... I am assuming there is some kind of race
condition going on, perhaps caused by slow or unreliable SIP phone
responses to call closedown events.

I looked at the zaptel code where this message is generated, in the
hope that I could request a flush of the channel that incorrectly
shows this channel open (if the telco is trying to put a call through,
then the line is definitely meant to be clear!) but it was way beyond
my ability to understand.

I thought about Glare (someone else suggested that in another
messsage), but the telco uses lowest-free channel, and we use Zap/1G,
so use the highest free channel.

Any thoughts or input are very welcome.

Thanks
Steve
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Re: [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk

[Steve Davies]

On 9/12/06, Olivier [EMAIL PROTECTED] wrote:

Hi,

What would you suggest to implement directed call pickup on bristuffed
Asterisk 1.2 ?
I'm after tle ability to pick a specific ringing call (without caring about
which call arrived first, for example).

 Something like : *8 + local extension would be perfect.

voip-info.org introduces many paths
(http://bugs.digium.com/view.php?id=5014 ,
http://linux.thorsten-knabe.de/asterisk/pickup.jsp, ...)
but which should be more stable ?


As far as I know, bristuff includes directed call pickup already,
using *8ext. Read their ChangeLog I think it has notes on how to use
it. If not, I am sure the list archives will have all the required
information.

Cheers,
Steve
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[asterisk-users] Problems getting 7970G upgraded to SIP

[Jason Lixfeld]

I have a 7970G with 5.0.3.0S Skinny (Load File: TERM70.5-0-3-0S) on  
it and I'd like to get it up to 8.x.


- With the SEPMAC.cnf.xml in place (which was taken from voip-info  
(http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP  
under This worked for me...)), I get Load ID Incorrect on the  
phone display after it boots.


The loadInformation line in the SEP file reads like this:

loadInformationSIP70.8-0-4SR1S/loadInformation

- If I remove the SEP file, the phone requests XmlDefault.cnf.xml.  I  
create the xml file based on the example from the same link above,  
the phone grabs the file, but doesn't upgrade.  It just sits in a  
loop of: release IP = renew IP = look for SEP, fail = look for  
XmlDefault, find and load XmlDefault = release IP...  The  
loadInformation line in the XmlDefault.cnf.xml file reads like this:


loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/ 
loadInformation6


- Here are TFTP server logs to illustrate that I'm using the correct  
case'd XmlDefault.cnf.xml file:


Sep 10 21:57:55 bubbles  tftpd[89195]: jalc7970.sip : read request  
for SEP00131A4D39F4.cnf.xml: File not found
Sep 10 21:57:55 bubbles  tftpd[89197]: jalc7970.sip : read request  
for //XmlDefault.cnf.xml: success


- All the files from the .cop are 100% unmodified.  I just tar -zxvf  
cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into  
the tftpd root directory, which is the same place the SEP and  
XmlDefault file are located.


Anyone have any ideas?

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Re: [asterisk-users] PRI: sometimes Asterisk drop calls

[Giorgio Incantalupo]

Hi,
thanks to all
I solved the calls dropped problem, it was resetinterval parameter in 
zapata.now asterisk does not drop calls anymore.

I do not get the message:

WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner

anymore...but I get all the others.
I'm interested to understand why I many messages like:

WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use 
on span 1.  Hanging up owner


How can a channel be already in use??? That means the channel is 
busy...if it is so then it is all right...but maybe that shouldn't be a 
warning but a notice or something else...should it?



TIA


Giorgio Incantalupo



Rich Adamson wrote:

Steve Davies wrote:

On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:

Steve Davies wrote:
 For the curious, can anyone tell me how this flag fixes the issue? 
- I

 have seen the error before, but always assumed it was related to hung
 channels.

 Thanks,
 Steve

 On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
 Problema solved!

 Just put resetinterval=never inside zapata.conf


 Giorgio Incantalupo

If memory serves correctly, I believe the parameter was added a couple
of years ago as a means / workaround for hung channels. At the time,
there was not any overwhelming evidence as why a channel would
occasionally hang. Some of the possibilities included unusual
interaction from the opposite end of the T1/E1, anomalies in the
dialplan, etc.

Now that a substantial amount of work / changes have been made relative
to PRI's and other internal asterisk code, there appears to be less 
of a

need to reset.

A reasonable approach might be to apply the parameter and pay close
attention to channels that might be in some strange state. If none are
observed, then leave it.


Thanks for that. I have a customer who is using Asterisk 1.0.x, and I
am tempted to backport this fix from the 1.2.x code where it was
introduced.


From a personal perspective, I think I'd hold off on the back port and 
devote that time towards testing the soon to be released version (now 
in Trunk).


If you've watched the number and type of changes that have gone into 
SVN Trunk in the last couple of months, it appears as though a 
significant number of possible memory leaks, sip code, infrastructure 
code, PRI code changes, etc, have been applied that would be 
beneficial for all production systems. There also appears to be a fair 
amount of work that will be needed to upgrade dialplan syntax (etc) 
for the new release.


Best guess is that once the Trunk code gets past the beta testing 
phase, it will likely be the asterisk code of choice for most/all 
production systems.


Consider the above is only my $0.02 worth. ;)

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Re: [asterisk-users] Rack for Asterisk with TDM2400 Digium board

[Raphael Jacquot]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Antoine Megalla wrote:
 Hi,
 
 I have a client who wants a call center with 16 analog
 FXO modules.
 I offered him a solution with a 1U or 2U rack and
 Digium TDM2400 card.
 I know that there mother board compatability issue
 with the Digium cards, so 
 can anyone suggest a barebone 1U or 2U server (I
 prefer the SuperMicro 
 Superservere series) with a Mother board that is
 compatible with the Digium 
 TDM2400 card (which is a Full Length PCI card).
 
 Thank you and best regards,
 
 Antoine Megalla 

avoid yourself the problems and go with tyan AMD64 motherboards...
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5-ecc0.1.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFBtS4Xqd/7Teiu2oRApY4AJ95UtnuJEqbWuIL+OYgSq8AdDPfywCcCrXs
GtOzanRSyTNQP3B84yPMsEA=
=s5KP
-END PGP SIGNATURE-
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Re: [asterisk-users] Dell hardware ...

[Warren (mailing lists)]

Alan Bunch wrote:
I was going to use a Dell 1425 for Asterisk build but I see on Digium's 
website that hardware may be problematic.  Can anyone shed a litle more 
light on the problem.   I see the Intel ethernet cards seem to cause 
problems.  If I need to disable the onboard Intel on the Dell hardware I 
can I just need to know what to expect.


How about the 850, any word there  ?


I am running trixbox 1.1.1 on an 850.  Actually on 2 850s, one is a 
hot-swap spare.  No problems.  The problems that were reported earlier 
on were with specific Digium boards on specific Dell machines.  I have a 
T-1 for my connectivity which comes in through a Cisco router and have 
no problems.


Look here for more info:
http://www.digium.com/en/docs/misc/compatibility_notes.php

W
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Re: [asterisk-users] Dropped call question - Maximum retries exceeded on transmission

[Dr. Michael J. Chudobiak]

Kohler, Jeffrey wrote:

I am encountering an intermittent issue where some of my calls are being
dropped.  Most of the calls that are made are successful.  However, some
calls will be dropped after having been connected for some time.

Each time a call gets dropped, I get output similar to the following in
the Asterisk console:

...

Does anyone have any suggestions?  I honestly don't know where to start
investigating this issue, so if anyone has any ideas they would be
greatly appreciated.


Jeffrey,

That's all a bit vague (how long before it drops, what protocol, are 
there firewalls, etc...), but my first guess would be a firewall NAT 
timeout. See the NAT Issues section at 
http://www.voip-info.org/wiki-IAX for example (it discusses IAX rather 
than SIP, but you get an idea of the issues).


- Mike
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Re: [asterisk-users] Rack for Asterisk with TDM2400 Digium board

[Matthew Fredrickson]

On Sep 12, 2006, at 7:32 AM, Antoine Megalla wrote:


Hi,

I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a barebone 1U or 2U server (I
prefer the SuperMicro
Superservere series) with a Mother board that is
compatible with the Digium
TDM2400 card (which is a Full Length PCI card).



The TDM2400P was designed in a way that it would not have all the 
motherboard compatibility issues that the other cards have had in the 
past.  As far as I know, you should be safe with pretty much any 
motherboard that you can fit it in.


Matthew Fredrickson

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Re: [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right

[Ricardo Carvalho]

Grandstream support just answered me saying that:
BT100/200 LCD does not supports alphanumeric caller ID display. You may 
want to try GXP-2000..

It's confirmed! Future firmwares won't support that feature! :(

Thanks to all that replied,

Regards,
Ricardo.







Craig Guy wrote:
The lcd in the current budgetone series cannot support alphnumeric 
display.


Craig

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

Sent: Tuesday, September 12, 2006 8:11 PM
Subject: Re: [asterisk-users] Grandstream Budgetone phones don't show



Thanks Jessee,

I've just sent an e-mail to Grandstream support asking if they are 
planning in a near future to release a firmware implementing 
alphanumeric callerid for Budgetone series.
When they answer me, I'll replay to this thread with their feedback, 
so the community can also benefit...


Regards,
Ricardo.






Jessee J Holmes wrote:

Ricardo,

From what I know its a physical limitation of the display 
Grandstream chose on that phone, Grandstream recommends purchasing 
the GXP-2000 phone instead if you're looking for this feature.


Grandstream has no plans from what I am aware of of making this 
change to the BudgetTone series phones.


You are more than welcome to inquire directly from Grandstream 
though, this is just from what I know from dealing with them in the 
past.



Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


Looking for voice over IP products?  Visit our VoIP store at 
http://voipstore.atacomm.com/




On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote:

I guess this functionality will be in the future added to new 
firmware releases don't you people think so?


Ricardo.






Doug Lytle wrote:


These phones aren't capable of alphanumeric entries, only numeric.

Doug


Tom Vile wrote:

They only do numeric callerid.





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Re: [asterisk-users] g729 problem

[Thomas Kenyon]

o o wrote:
 
 
 Thomas,
Thanks for your help so far. I finally figured out
 where 'debug level 10' dumps to. In reading the logs
 there, it's telling me I'm out of licenses. I'm not a
 math wizard by any means, but I would assume with g729
 on the GXP-2000 and on the IAX trunk, I would only
 need 1 license to transcode my IVR prompts to the
 incoming caller.

Well, first-off it would be a good idea to re-encode them to g.729.
If you are using trunk then there is a convert tool for this, failing
that you can use the tool on.

http://www.asteriskguru.com/tools/audio_conversion.php

The ast-linux site can provide you with all the default prompts
re-encoded (well, re-recorded then encoded).

 However, it seems to use all 6
 available, and never releases them, even after hanging
 up the call. I haven't found a way to see what
 process(s) are using each license instance.

If you are using monitor/mixmonitor or a meetme room you will run out of
licenses very quickly as both applications will need to transcode the
stream to slin and bakc again.

 I
 downloaded the re-recorded set you referenced, and I
 can hear the default system prompts, but all my
 previously recorded prompts are null because of the
 'out of license' issue. For the recording, even with
 console verbosity set to 16, the out of license
 messsage was never logged to the console,

Strange, maybe this behaviour changed in later versions (It's been a
while shice it happened to me).

 only to the
 debug text. I'm off to find a way to transcode my
 custom prompts into g729 (or get the freepbx recording
 interface to do so for new prompts)

See above.

 but if someone can
 help me determine why * thinks I need more than 6
 licenses for a single incoming call I would appreciate
 it. TIA

Which applications are running alongside the call?

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Re: [asterisk-users] Dell hardware ...

[Steve Rawlings]

Hi Arjan,

We're thinking about purchasing a Dell 1850 for a new production Asterisk, 
could you detail your spec, ie processor, memory, raid or whatever, it could 
really help me.  We too have a 4 port Digium PRI, a TE405 and also a TDM22b. 
I know our requirements could be different from yours, we're looking at 
about 70 SIP users, including maybe 35 agents with calls recorded 
eventually, but would be good to have a baseline to work from, many thanks.


Regards,

Steve


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

Sent: Tuesday, September 12, 2006 3:14 PM
Subject: RE: [asterisk-users] Dell hardware ...


Hi, Alan,

We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
in it and it works perfect.
It is almost PlugPlay.

greetings


Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: dinsdag 12 september 2006 8:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dell hardware ...

Hi Alan,
simply do not use Dell hardware.
we had your problem, we called Dell and they told us that our servers
was not configurable (they were too cheap).
So now I do not use Dell anymore and we have less problem.


Giorgio Incantalupo


Alan Bunch wrote:

I was going to use a Dell 1425 for Asterisk build but I see on
Digium's website that hardware may be problematic.  Can anyone shed a
litle more light on the problem.   I see the Intel ethernet cards seem



to cause problems.  If I need to disable the onboard Intel on the Dell



hardware I can I just need to know what to expect.

How about the 850, any word there  ?

TIA

Alan




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Re: [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains

[Dave Cotton]

On Tue, 2006-09-12 at 09:00 -0600, [EMAIL PROTECTED] wrote:
 I've ever post this question many times on asterisk
 users without success ?

As I and many others have probably noted.

I found then neatly filed in junk mail.

Perhaps you're getting your just deserts.

-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [asterisk-users] WG: Asterisk and Agents

[Ira]

At 06:43 AM 9/12/2006, you wrote:

It's not a great answer, but since it's only a problem adding you 
might just have to validate the codes the agents type in.



exten = _*8XXX,1,Answer


exten = _*8XXX,n,gotoif($[${EXTEN:1}  8032]?GoodOne)
exten = _*8XXX,n,goto(hangup)


exten = _*8XXX,n(goodone),SetLanguage(de)
exten = _*8XXX,n,AddQueueMember(DEMO|Agent/${EXTEN:1})
exten = _*8XXX,n,Dial(Local/999/n,,D(#))
exten = _*8XXX,n,AgentCallBackLogin(${EXTEN:1}|[EMAIL PROTECTED])
exten = _*8XXX,n(hangup),Hangup()


That still doesn't solve an agent putting in the wrong number if it's 
valid, but it limits it to valid entries.  You could add this to tell 
them what they entered:.


exten = saydigits(${EXTEN:2})

That won't say the 8 which shortens the messages and gives a better 
chance they might listen.


Ira 


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[asterisk-users] Polycom MyStat

[Douglas Garstang]

Has anyone ever gotten the Polycom MyStat soft-key to do anything?

Setting the status to something like 'Away', does not generate any outgoing SIP 
traffic from the phone. Calling into the phone either from a watched buddy, or 
other number, acts as if the status was never changed. A call to Polycom 
yielded no results. 

Thanks,
Doug.
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[asterisk-users] Verizon ISDN service in NY Hunt Groups

[Bernie Courtney]

Title: Verizon ISDN service in NY  Hunt Groups






Is anybody on here using PRI or BRI service in New York state with the trunks in a hunt group from Verizon??

I'm trying to setup a system and I've spoken to three people at verizon who all claim they cant put BRI or PRI circuits into a hunt group, I find that EXTREMELY hard to believe.

If you've had success could you share the person you spoke with and or what you asked for (or better yet a tarriff #)

thanks!
Bernie



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[asterisk-users] Calling Card and Billing

[[EMAIL PROTECTED]]

Hi Users,Im looking for recommendations on softwares for calling card implementation and post paid billing services. Please give some recomendatons based on your experiences. A detail of the pros or cons (if any) you faced with it would be highly welcome. 
thanks in advance.Dan
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[asterisk-users] A simple goal, help me please!

[David R.]

Okay. I'm setting up my first Asterisk box and the only thing I want to do right now is get my Ekiga softphone to register with it. Here is how I have my sip.conf set up:-sip.conf-[general]context=default
srvlookup=yes[davidr64]type=friendsecret=welcomequalify=yesnat=nohost=dynamiccanreinvite=nocontext=internal-/sip.conf-I have my softphone (on another box) set up with this info:
Account Name: TestingRegistrar: 10.20.30.71 (correct IP for the asterisk box)User: davidr64Password: welcomeAuthentication Login: davidr64Realm/Domain: 
10.20.30.71Registration Timeout: 3600

That should be all I need, right? When I tell my softphone to register, my Asterisk console shoots this at me:Sep 12 11:38:50 NOTICE[1799]: chan_sip.c:10886 handle_request_register: Registration from '
sip:[EMAIL PROTECTED]' failed for '10.20.30.48' - Wrong passwordIs there something I'm missing? If not, what do I have wrong?Thanks,David
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[asterisk-users] Re: Calling Card and Billing

[[EMAIL PROTECTED]]

Hi all,Let me add to my query.I would prefer to have an Asterisk GUI that has billing  calling card solution all in one.Got any suggestions.?Thanks
On 12/09/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Users,Im looking for recommendations on softwares for calling card implementation and post paid billing services. Please give some recomendatons based on your experiences. A detail of the pros or cons (if any) you faced with it would be highly welcome. 
thanks in advance.Dan


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[asterisk-users] test

[harrygaillac-sip]

test








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Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos 
expériences. 
http://fr.answers.yahoo.com 

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[asterisk-users] AEL if/else/IFTIME fun.

[Dan Serban]

I've been playing a lot with AEL and one thing seems to be perplexing me,
it's regarding actions taken using the IFTIME command, and they don't seem
to be having much affect.

Here's my AEL script:

// Main menu configuration

context mainmenu {
includes {
default;
};

1 = goto canadamenu|s|1;
2 = goto usmenu|s|1;
3 = agentqueue(mx);
4 = goto adminmenu|s|1;

// T1 PRI
2977 = goto s|1;

s = {
Ringing();
Wait(1);
Set(attempts=0);
Answer();
Wait(1);
if( $[IFTIME(06:00-11:59,mon-fri,*,*) |
IFTIME(07:00-11:59,sat,*,*) | IFTIME(08:00-11:59,sun,*,*)]) {
// Morning
Background('menu/tlc-good-morning');
}
if( $[IFTIME(12:00-16:59,*,*,*)] ) {
// Afternoon
Background('menu/tlc-good-afternoon');
}
if( $[IFTIME(17:00-20:00,mon-fri,*,*) |
IFTIME(17:00-18:00,sat-sun,*,*) ] ) {
// Evening
Background('menu/tlc-good-evening');
}
else {
Background('menu/tlc-after-hours');
WaitExten(5);
goto default|200|1; // General Mailbox
Hangup();
};
repeat:
// Main
Set(attempts=$[${attempts} + 1]);
Background('menu/tlc-home-menu');
WaitExten(5);
if( ${attempts}  2 ) goto repeat;
adminqueue(operator);
Hangup();
};
};

The above is fairly self explanatory, and based on what I could glean
through googling, it should be correct.  Though I've tried different
variations on the above implementation of the IFTIME calls.  (using else if,
IFTIME by itself etc.)

What happens, is that it only plays the good morning greeting at any time
during the day.  What am I doing wrong?
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Re: [asterisk-users] Verizon ISDN service in NY Hunt Groups

[Doug Lytle]

Bernie Courtney wrote:


I'm trying to setup a system and I've spoken to three people at 
verizon who all claim they cant put BRI or PRI circuits into a hunt 
group, I find that EXTREMELY hard to believe.




PRIs don't use hunt groups (Just found this out myself).  An inbound 
phone number will take up as many channels as available on the PRI for 
each person calling.  I've had to take this into account on my inbound 
fax numbers and limit each call using the PRI_CAUSE=17 (User busy) to 
limit to 1 call.


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[asterisk-users] Trouble connecting to my telco with fonebridge

[Leif Kristian Hetlesæther]

Hi list

I'm a bit stuck connecting my fonebridge from redfone to my telco. I
have a E1 line (30 channels).

The configuration is sent to the fonebridge and the led lights up red
and stays that way. I am using CentOs 4.4 and zaptel . 1.2.9.1 (had the
same problem with CenOs 4.3 zaptel 1.2.6. Also tried fc5)

Any suggestion on how to trouble shoot?

My provider is Priority telecom Norway. Anyone having a working config?

Underneath my redfone.conf and zaptel.conf

Rgards
Leif Hetlesæther


redfone.conf


span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,2,0,ccs,hdb3,crc4
card=eth0
source=00:08:02:ED:4B:F8
destination=00:0C:42:03:5C:8A


zaptel.conf

dynamic=eth,eth0/00:0C:42:03:5C:8A/0,31,1
bchan=1-15
dchan=16
bchan=17-31
# Global data
loadzone= no
defaultzone= no

---





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[asterisk-users] RE: [asterisk-biz] Come see us at VON

[Dean Collins]

On a similar note, there is a get together 6-8pm on Wednesday evening in
Room 211, it's open to anyone involved with Asterisk, if you have any
questions Carl Ford is the contact.

I'm just reposting as I haven't seen many emails about this get together
and wanted to make sure everyone knew.

I know for a fact there will be some interesting things to see :)


Regards,
 
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-biz-
 [EMAIL PROTECTED] On Behalf Of Lonnie Lazar
 Sent: Tuesday, 12 September 2006 1:21 PM
 To: asterisk-biz@lists.digium.com
 Subject: [asterisk-biz] Come see us at VON
 
 If you're in Boston between September 12 - 14 make sure to come by
 the Voxilla booth to say hi; #1462.  We've got some interesting new
 total IP Communications solutions to tell you about, and a VON Show
 special on the Linksys WIP300.  You can order on the web at http://
 store.voxilla.com and remember, Asterisk users always get a discount
 on everything we sell at Voxilla by using the *user coupon code at
 check-out.
 
 All the best,
 --
 Lonnie Lazar
 V.P. Sales
 Voxilla, Inc.
 [EMAIL PROTECTED]
 http://store.voxilla.com
 http://www.voxilla.com
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Re: [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2 Asterisk

[Olivier]

2006/9/12, Steve Davies [EMAIL PROTECTED]:
As far as I know, bristuff includes directed call pickup already,using *8ext. Read their ChangeLog I think it has notes on how to useit. If not, I am sure the list archives will have all the requiredinformation.
Cheers,SteveReading http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffPickUpChan
 and bristuff changelogs, I couln't know whether :1. PickUpChan really provided directed pickup on non-Snom SIP hardphones at it appears to pickup the last incoming call, no matter the extension you asked to be picked,
2. PickUpChan was deprecated in favor of fully-backed Asterisk standard code as various Mantis bugs suggest.Regards
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[asterisk-users] Polycom Soundpoint Key Remap

[Shawn Kelley]

Im told by Adam below that I can use a Speed Dial to accomplish this.However, I dont know how to map a speed dial to the key.I know the syntax for mapping a function to it ( IP_500 key.IP_500.31.function.prim=BuddyStatus/ )However, I dont know how to do a speed dial.Any one out there know?Thanks!--ShawnShawn Kelley wrote: Hi, Does anyone know how to do a re-map of a key on the Polycom to make it  dial a number. I know how to remap a key to a certain function, but I dont know how  to make it dial a number. Im wanting to re-map the Service key to dial *8 for a group pickup. Any help is greatly appreciated. Thanks! --ShawnAFAIK, you will need to tell it save a speed dial for *8, and then map the key to dial the speed dial number that you saved it as.Hope this helps.Regards,Adam








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[asterisk-users] Please help with a telular mod. SX5e

[Jorge Cisneros]

Hi I have 5 telular mod SX5e https://www.telular.com/v2/html/products/product_display.asp?productID=94 is a great stuff, but i have a extrange problem with asterisk. Sometimes the sound is ugly or choppy. The telular alone work fine all the time
 For example if i made 5 calls from asterisk to gsm network, but 2 or 3 calls the sound is really bad but only in the side of asterisk. I try all the echo cancellers but nothing work. I have 2 setups
 sipphone == asterisk with channel bank == telular SX5e == GSM network sipphone == asterisk with wctdm 4FXO == telular SX5e == GSM networkIn both case is the same.
Any idea? do you have a similiar problem?Thanks
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Re: [asterisk-users] Problems getting 7970G upgraded to SIP

[Richard Klingler]

Hi Jason


loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/loadInformation6


1. Stick with the 8.0.2 SIP image as it works best with asterisk...
   at least for me (o;



- Here are TFTP server logs to illustrate that I'm using the correct 
case'd XmlDefault.cnf.xml file:


Sep 10 21:57:55 bubbles  tftpd[89195]: jalc7970.sip : read request for 
SEP00131A4D39F4.cnf.xml: File not found


2. I thought you created your SEP file? And still it can't be found?

Sep 10 21:57:55 bubbles  tftpd[89197]: jalc7970.sip : read request for 
//XmlDefault.cnf.xml: success


3. Wondering what messages are coming after that...or is it the
   point where it starts over again?



- All the files from the .cop are 100% unmodified.  I just tar -zxvf 
cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into the 
tftpd root directory, which is the same place the SEP and XmlDefault 
file are located.


4. So you have all those:

-bash-2.05b$ tar tzvf cmterm-7970_7971-sip.8-0-2SR1.cop
 644 Mar 22 23:49 SIP70.8-0-2SR1S.loads
 2538161 Mar 22 23:49 apps70.1-1-1-15.sbn
  411264 Mar 22 23:49 cnu70.3-1-1-15.sbn
1996 Mar 23 00:06 copstart.sh
 2401588 Mar 22 23:49 cvm70sip.8-0-1-18.sbn
  483105 Mar 22 23:49 dsp70.1-1-1-15.sbn
  465288 Mar 22 23:49 jar70sip.8-0-1-18.sbn
  71 Mar 23 00:06 load119.txt
  72 Mar 23 00:06 load30006.txt
   0 Mar 23 00:06 signed/
 4046848 Mar 23 00:06 signed/cmterm-7970_7971-sip.8-0-2SR1.cop
 644 Mar 22 23:49 term70.default.loads
 644 Mar 22 23:49 term71.default.loads



Anyone have any ideas?


5. Not yet. But might be you need to go with a firmware
   in between first before going with 8.0.x.



cheers
rick


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Re: [asterisk-users] Polycom Soundpoint Key Remap

[Noah Miller]

Hi Shawn -

Unfortunately, on a Polycom, you can no longer remap a speed dial to a
key.  You can set extra line appearances to be speed dials (I can show
you that, if you want), but none of the other keys.  This feature used
to be available, but was quietly removed as of 1.5.x.  If you want to
revert to 1.4.1 you can do it with the subpoint feature (I can show
you that, too), but 1.4.1 has other serious limitations.

- Noah



On 9/12/06, Shawn Kelley [EMAIL PROTECTED] wrote:



I'm told by Adam below that I can use a Speed Dial to accomplish this.
However, I don't know how to map a speed dial to the key.
I know the syntax for mapping a function to it ( IP_500
key.IP_500.31.function.prim=BuddyStatus/ )
However, I don't know how to do a speed dial.

Any one out there know?

Thanks!
--Shawn



Shawn Kelley wrote:

 Hi,

 Does anyone know how to do a re-map of a key on the Polycom to make it
 dial a number.

 I know how to remap a key to a certain function, but I don't know how
 to make it dial a number.

 I'm wanting to re-map the Service key to dial *8 for a group pickup.

 Any help is greatly appreciated.

 Thanks!

 --Shawn

AFAIK, you will need to tell it save a speed dial for *8, and then map
the key to dial the speed dial number that you saved it as.

Hope this helps.

Regards,
Adam



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Re: [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry

[Noah Miller]

I note that the SC420 is listed as incompatible but the SC430
appears to be a slightly different beast in terms of chipset, the 430
has the newer E7230 as opposed to the E7221 - does this make a
difference to compatibility?


I have an SC420 in one office that works quite well.  I think the
confusion comes because you can't use the bios to change what
interrupts are used by the various integrated components on the
server.  The result is that if you use the first PCI slot, the Digium
card WILL share an interrupt with the integrated network interface,
and you're liable to get missed interrupts and choppy sound under
heavy network traffic.  You can easily get around this by using a
different PCI slot.  I have perfect sound with the SC420 (although
I've never heavily taxed this machine).

- Noah

On 9/9/06, Gunnar Schaller [EMAIL PROTECTED] wrote:

Hello Matthew,
It depends on the chipset on the mainboard. I had problems with a
SC1420, the only way to solve it was to get a new server (without
Intel chipset). So don't try a chipset which is listed on the Digium
compatibility site.



Wednesday, September 6, 2006, 8:55:58 AM, you wrote:

 We're looking at using a number of Dell Poweredge SC430 servers as
 Asterisk hosts in our smaller overseas offices with Digium cards in
 to provide local breakout over the pre-existing analogue or digital
 phone lines (One office uses ISDN2 the others analogue)

 I note that the SC420 is listed as incompatible but the SC430 appears
 to be a slightly different beast in terms of chipset, the 430 has the
 newer E7230 as opposed to the E7221 - does this make a difference to
 compatibility?

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[asterisk-users] consitent half channel loss after 6 minutes

[Jerry Geis]

I have a TDM2402E card calling out to NuFone using IAX2
and after couple of minutes (6 min to be exact) the recipient
can no longer hear the caller.
The caller can continue to hear the recipient clearly.

After the 6 min I continued to listen to the call and I a could
hear the other person but they could not hear me at all. After
some time we gave up and hung up. The channel never came back.

How can I find out the issue here? When the channel was lost
I tried toggling the Hold button and the other user heard a few seconds
of my voice. But coming back off hold it was no longer being heard still.

I dont think this is nufone. Something is not right how do I go about
finding what is wrong? Any suggestions.

I am using asterisk-1.2.11 and zaptel-1.2.8.

Jerry
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[asterisk-users] strange problem with calls between MGCP and SIP clients(ATA's)

[Andy Kuo]

Hi,

We have experience problems with calls between MGCP ATA's and SIP
ATA's (Linksys PAP2-NA).
A call from MGCP or SIP to the other connects normally and the
conversation can usually last around 30 seconds and it becomes one-way
audio.

What I don't understand is how the calls can be set up and talk for a
few seconds without problems and suddenlly go wrong.  If there are
problems, such as misconfiguration, the call should not even be
connected, or at least the on-way audio problem should start right
from the beginning, shouldn't it?

I know MGCP is not very popular here, but we have quite a few of them
on hand that we would really like to use.
Any comments/suggestions are greatly appreciated.

Thanks.
Andy
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Re: [asterisk-users] How to setup announce attibute in queues.conf

[Artifex Maximus]

Hello,

announce = support-department
plays support-department.wav so playing support-department-recording.wav needs
announce = support-department-recording

bye,
Zsolt

On 9/12/06, gc [EMAIL PROTECTED] wrote:


I have this line in my queues.conf:
announce= support-department
and I have an recording file
support-department-recording.wav file.
Can anybody tell me how to setup support-department so it play the .wav file
when agent pickup the phone? Where should I define support-department so
asterisk will play support-department-recording.wav? Is
this in musiconhold.conf?

gc

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RE: [asterisk-users] Polycom Soundpoint Key Remap

[Douglas Garstang]

The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it yesterday 
on 2.0.1

 -Original Message-
 From: Noah Miller [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, September 12, 2006 1:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap
 
 
 Hi Shawn -
 
 Unfortunately, on a Polycom, you can no longer remap a speed dial to a
 key.  You can set extra line appearances to be speed dials (I can show
 you that, if you want), but none of the other keys.  This feature used
 to be available, but was quietly removed as of 1.5.x.  If you want to
 revert to 1.4.1 you can do it with the subpoint feature (I can show
 you that, too), but 1.4.1 has other serious limitations.
 
 - Noah
 
 
 
 On 9/12/06, Shawn Kelley [EMAIL PROTECTED] wrote:
 
 
  I'm told by Adam below that I can use a Speed Dial to 
 accomplish this.
  However, I don't know how to map a speed dial to the key.
  I know the syntax for mapping a function to it ( IP_500
  key.IP_500.31.function.prim=BuddyStatus/ )
  However, I don't know how to do a speed dial.
 
  Any one out there know?
 
  Thanks!
  --Shawn
 
 
 
  Shawn Kelley wrote:
  
   Hi,
  
   Does anyone know how to do a re-map of a key on the 
 Polycom to make it
   dial a number.
  
   I know how to remap a key to a certain function, but I 
 don't know how
   to make it dial a number.
  
   I'm wanting to re-map the Service key to dial *8 for a 
 group pickup.
  
   Any help is greatly appreciated.
  
   Thanks!
  
   --Shawn
  
  AFAIK, you will need to tell it save a speed dial for *8, 
 and then map
  the key to dial the speed dial number that you saved it as.
 
  Hope this helps.
 
  Regards,
  Adam
 
 
 
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[asterisk-users] Switch Experiences

[Ben Gore]

Hello:

I'm would like to get feedback before finalizing design of a VOIP 
network, in particular about people's experience with network (primarily 
10/100/1000 twisted pair) ethernet switches.


I have a number of candidates in mind, but I would like any and all 
opinions and suggestions on the following topics:


-Throughput/minimal latency/delays;
-Managed vs unmanaged;
-Redundant links/auto healing;
-Redundant power supply;
-Configuration of port attributes (i.e. locking 10 M/b interface to 10 
M/b instead of leaving in AUTO);

-Resistance to Electrostatic/Electromagnetic/RF energy;
-Shielded vs unshielded ports  cables;
-Pricing;
-Any other relevant information.

The reason for asking is there seems to be a significant amount of 
disagreement about a number of these issues from a variety of experts, 
while there's a considerable amount of experience on this list in these 
areas.


Suggestions of specific manufacturers and models welcome if you've had 
good luck with them.


-Ben



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[asterisk-users] sip origination and termination

[Christopher Corn]

im finding companies like voxee that offer very low rates and then companies like voipstreet that offer at a higher rate doube. whats the catch? is voxee, what you would call a wholesaler and voipstreet, commercial?im worried about going with companies like voxeee, because i question their support. what are your guys thoughts on this? Thanks.___
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RE: [asterisk-users] Polycom Soundpoint Key Remap

[Watkins, Bradley]

Did you ever try to get it working on any 1.6.x releases?  I hacked at
it a bit and it didn't seem to be working, though I could have been
doing something wrong.  I was, after all, reading the manual... ;)

I'm glad to hear someone successfully doing it, as it's something I've
wanted to play with for awhile now.

- Brad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, September 12, 2006 4:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom Soundpoint Key Remap

The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it
yesterday on 2.0.1

 -Original Message-
 From: Noah Miller [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, September 12, 2006 1:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap
 
 
 Hi Shawn -
 
 Unfortunately, on a Polycom, you can no longer remap a speed dial to a

 key.  You can set extra line appearances to be speed dials (I can show

 you that, if you want), but none of the other keys.  This feature used

 to be available, but was quietly removed as of 1.5.x.  If you want to 
 revert to 1.4.1 you can do it with the subpoint feature (I can show 
 you that, too), but 1.4.1 has other serious limitations.
 
 - Noah
 
 
 
 On 9/12/06, Shawn Kelley [EMAIL PROTECTED] wrote:
 
 
  I'm told by Adam below that I can use a Speed Dial to
 accomplish this.
  However, I don't know how to map a speed dial to the key.
  I know the syntax for mapping a function to it ( IP_500 
  key.IP_500.31.function.prim=BuddyStatus/ ) However, I don't know 
  how to do a speed dial.
 
  Any one out there know?
 
  Thanks!
  --Shawn
 
 
 
  Shawn Kelley wrote:
  
   Hi,
  
   Does anyone know how to do a re-map of a key on the
 Polycom to make it
   dial a number.
  
   I know how to remap a key to a certain function, but I
 don't know how
   to make it dial a number.
  
   I'm wanting to re-map the Service key to dial *8 for a
 group pickup.
  
   Any help is greatly appreciated.
  
   Thanks!
  
   --Shawn
  
  AFAIK, you will need to tell it save a speed dial for *8,
 and then map
  the key to dial the speed dial number that you saved it as.
 
  Hope this helps.
 
  Regards,
  Adam
 
 
 
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Re: [asterisk-users] sip origination and termination

[broadbandvoice]

You're right Voxee support sucks. But I think they do well and provide good rates. I'm using Gafachi, a little expensive and have Voxee. I'm using LCR so the termination will try Voxee first and when not available will use Gafachi. You can set up something like that with a least cost routing.

-- Original message -- From: Christopher Corn [EMAIL PROTECTED] 
im finding companies like voxee that offer very low rates and then companies like voipstreet that offer at a higher rate doube. whats the catch? is voxee, what you would call a wholesaler and voipstreet, commercial?

im worried about going with companies like voxeee, because i question their support. 

what are your guys thoughts on this? Thanks.

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Re: [asterisk-users] Polycom Soundpoint Key Remap

[Noah Miller]

Hi Doug -


 AFAIK, you will need to tell it save a speed dial for *8,
 and then map
 the key to dial the speed dial number that you saved it as.

The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it
yesterday on 2.0.1


I just noticed that version 2.01 came out.  I'm really glad to hear
you got it working.  I've wanted to use that feature for a long time.

If you can make it work on 1.6.x or 1.5.x, can you send me a config?
It does not work according to the manual.  When I called Polycom they
said at the time (pre 2.x) they no longer supported that feature.

Thanks!
Noah
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[asterisk-users] sound file length

[Raphael Jacquot]

At some point in my dial plan, I need to find out the length of a sound 
file in seconds (to weed out things that are way too short)


the record application doesn't seem to have any facilities to do that.

any ideas ?
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[asterisk-users] All circuits are busy now???

[BerkHolz, Steven]

"All 
circuits are busy now" makes perfect sense in my 
PRI trunk is full.

How do I stop 
asterisk from playing this recording when it is a wrong/bad 
number?

I gat a call 
today that a user was trying "all day" to call a number in Mexico and kept 
getting the above recording.

I said, try in 
on your cell phone, and they received a "this number is not is 
service".

I would like 
to either hear the far recording (I think I will get billed for this), or 
internally play a different message.

I think the 
issue is that I am using a PRI and am receive the cause code that is triggering 
the above recording.

Can asterisk 
play a different message for this? and only play the above message if "MY" 
circuit is busy?




Thank You,
Steven 
BerkHolz- MCSA 
- MCSE -Manager of Information SystemsTESCO Group 
CompaniesFax. 248-836-5101www.TESCOGroup.com
Board member 
ofwww.glimasoutheast.org

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Re: [asterisk-users] Rack for Asterisk with TDM2400 Digium board

[C F]

Even though it might work, you should realy consider using Quad span
T1s with channel banks, or Xorcoms Astribank solutions.

On 9/12/06, Matthew Fredrickson [EMAIL PROTECTED] wrote:


On Sep 12, 2006, at 7:32 AM, Antoine Megalla wrote:

 Hi,

 I have a client who wants a call center with 16 analog
 FXO modules.
 I offered him a solution with a 1U or 2U rack and
 Digium TDM2400 card.
 I know that there mother board compatability issue
 with the Digium cards, so
 can anyone suggest a barebone 1U or 2U server (I
 prefer the SuperMicro
 Superservere series) with a Mother board that is
 compatible with the Digium
 TDM2400 card (which is a Full Length PCI card).


The TDM2400P was designed in a way that it would not have all the
motherboard compatibility issues that the other cards have had in the
past.  As far as I know, you should be safe with pretty much any
motherboard that you can fit it in.

Matthew Fredrickson

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Re: [asterisk-users] All circuits are busy now???

[Eric \ManxPower\ Wieling]

BerkHolz, Steven wrote:
 


All circuits are busy now makes perfect sense in my PRI trunk is full.

 

How do I stop asterisk from playing this recording when it is a 
wrong/bad number?


 

I gat a call today that a user was trying all day to call a number in 
Mexico and kept getting the above recording.


 

I said, try in on your cell phone, and they received a this number is 
not is service.


 

I would like to either hear the far recording (I think I will get billed 
for this), or internally play a different message.


 

I think the issue is that I am using a PRI and am receive the cause code 
that is triggering the above recording.


 

Can asterisk play a different message for this? and only play the above 
message if MY circuit is busy?


Sounds like you are using some Asterisk GUI.  Can't help with that and 
this message will only be useful to others.


When Dial exits it will set the value of HANGUPCAUSE to something.  Use 
the dialplan to play different messages depending on the value of 
hangupcause.


See show application dial in the Asterisk CLI, 
/path/to/src/asterisk/docs/README.variables, 
http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf 
for the cause code values, and 
/path/to/src/asterisk/include/asterisk/causes.h for which causes 
Asterisk knows about.



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[asterisk-users] INX (internationalnumber.com) Outgoing problem

[Daniel Cyt]

Dear friends,
I'm trying to dial out using a INX (internationalnumber.com) line but I get the message "ths account number is not valid".My asterisk is working well with other providers. INX support told me the line is working and in fact when I setup this line on my softphone it works.
I was googling and Isaw some people with the same problem as I'm having now on the list but I could not find any solution or answer for those questions.
Please, does anyone know what could be wrong?
Here is my sip.conf and my extensions.conf: http://pastebin.ca/168384Thank you very much for your help and attention
DanielMSN Messenger: converse com os seus amigos online. Instale grátis. Clique aqui. 

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[asterisk-users] Virtualise asterisk on Xen

[Arik Raffael Funke]

Hi,

has anybody experience running asterisk on a (i.e. fedora-based) Xen 
system? What about mISDN support etc.?


For a low-load system I thought about using:
1. Sempron 2800+
2. some memory, in your opinion how much should I attribute to the 
asterisk guest system?

3. A AVM Fritz!PCI card for PSTN access
4. HFCPCI-S card in nt-mode for internal ISDN bus provision
5. Asterisk 1.2 with chan_misdn for the ISDN-card support

It would be great to hear some of your thoughts on this set-up?

Regards,
Arik


NB: I have the impression that virtualisation is not a big issue on this 
mailing list... Is that due to a show-stopper I overlooked, just because 
everything goes so smoothly that nobody even bothers to mention it ;-), 
or because everybody has plenty of hardware they can dedicate to their PBXs?


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[asterisk-users] Makefile.moddir_rules: No such file or directory

[Ronald Wiplinger]

I need h.264 and tried therefore svn checkout 
http://svn.digium.com/svn/asterisk/trunk asterisk


(currently I have branches 1.2 installed)


make clean; make update; make install

.

make[1]: Entering directory `/usr/local/src/svn-versions/asterisk'
rm -f .depend
rm -f .depend
rm -f .depend
Makefile:60: /usr/local/src/svn-versions/asterisk/Makefile.moddir_rules: 
No such file or directory
make[2]: *** No rule to make target 
`/usr/local/src/svn-versions/asterisk/Makefile.moddir_rules'.  Stop.

make[1]: *** [channels-clean-depend] Error 2
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk'
make: *** [update] Error 2


Why is  Makefile.moddir_rules missing, or what have I forgotten to do?

bye

Ronald
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[asterisk-users] Bad number - is not in inbound speed dial

[Enrico Pasqualotto]

Hi, what mean this voice message that asterisk say when I try to call an
extension of another asterisk connected by IAX2 trunk?

This problem exist only if I call from asterisk1 to asterisk2, vice
versa all work.
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Re: [asterisk-users] sound file length

[Time Bandit]

At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)

the record application doesn't seem to have any facilities to do that.

any ideas ?

use sox beep.wav -e stat and parse the output

man is your friend
google also :)
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