date:20060909


Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

---BeginMessage---
Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
 Tim, this is the way I have Gafachi set up in sip.conf and works well with
 channels that have an ATA attached to it but not the virtual one. I have
 changed the host in extensions.conf to the .sip.gafachi.com.
 But I have calls on the server and cannot restart it yet. I'll keep you
 posted and thanks for the feedback.

 [gafachi-out]
 type=peer
 secret=xx
 username=x
 fromuser=x
 fromdomain=xxx
 host=.sip.gafachi.com
 ;usereqphone=yes; This provider requires ;user=phone on
 URI ;nat=yes
 rtptimeout=60
 dtmfmode=rfc2833


 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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[asterisk-users] Call Processing Slow 11 seconds


I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3

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RE: [asterisk-users] What don't I get about SIP?

2006-09-09 Thread Mike

 
 Actually, as soon as you hit 8 you will get the fast busy.
 
 Is that your full dialplan? What about an emergency (911) or 
 other N11 calls? What about direct dial international calls (011...)?

Its my full test dialplan for now.  I do get fast busy as soon as I hit 8,
so that part works.


 So, it looks like what you want is a global dialing timeout 
 in the phone, which the Polycom phones don't appear to have 
 once you break dialtone. But you may be able to kluge the 
 digit timeout to give you that feature if you don't need it 
 for what it is meant for. Right now you are using it to 
 timeout when a digit other than 1 is pressed after the 9. 
 That isn't really necessary (unless 91 followed by 9 digits 
 is actually a valid number for whatever you are doing with 
 it). Also, you are using the brackets unecessarily, since you 
 only have one digit within them. An equivalent dialplan that 
 doesn't use the digit timeout feature would be:
 
 digitmap dialplan.digitmap=7xx|9[2-9]x|91xx

Fair enough, but that doesn't solve the original issue, but it makes my
kludge a bit better

[lots of good info removed]

 Note that asterisk may possibly respond with error code 484 
 if the sequence pressed isn't complete, which would make the 
 phone continue to ask for more digits. So, the other part of 
 the solution is to add:
 
 exten = _X.,1,Congestion()
 That will match anything that doesn't match one of your valid 
 extensions as long as it is two digits or more. So you still 
 will get the behaviour you don't like if someone just presses 
 7 or 9 and nothing else. But it will give you most of what 
 you want, assuming I understand what you are looking for in 
 the first place (you could try x.T in the digitmap and _. 
 in extensions.conf, but _. is likely to cause other problems).

Did I misread the Asterisk wiki pages, because I believed that when a
pattern was present, the pattern takes precedence over any real
extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)?


Thanks John, I appreciate all the info.


Mike

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[asterisk-users] Another (quick) Polycom 501 question

2006-09-09 Thread Mike




Hi 
all,

That's my last one 
for a while (I hope).

How can I (if at all 
possible) make the 501 turn on the speaker phone as soon as a digit is dialed 
(if the handset is not lifted)?Sort of likewhat a normal 
speakerphone does.

The reason I want this is I want the 501 digitmap to be taken into 
consideration even if the handset isnt lifted and the speakerphone button isn't 
consciously pressed. For all those users who don't want to press send, but 
like dialing without lifting the handset (and can't be bothered to press the 
speakerphone button). Yes I know it's capricious, but we have the users we 
have...

Yes, I have read the 
admin manual, but couldn't find the info. I am assuming I just don't know 
what to look for, but that this functionality exists.



Mike
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Re: [asterisk-users] Stupid question about FXS/FXO

2006-09-09 Thread Moises Silva


Hi Ivan. As you see in this page:

http://www.neobits.com/do/dtls?pid=9583

This card is a bundle,  wich means supports both, FXO, and FXS. FXO
ports should be used to connect your 3 telco lines, and FXS port to
connect some phones.

Regards


On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote:

Hi yet again,

Is a TDM13B card the correct one if I have three phone lines and I
want to use the extra port to connect a normal phone?

Thanks!
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Re: [asterisk-users] Stupid question about FXS/FXO

2006-09-09 Thread Iván Vega R.


Yeah I figured after some experimentation :) Thanks.

I got confused while configuring zaptel and asterisk, hehe.


On 9/9/06, Moises Silva [EMAIL PROTECTED] wrote:

Hi Ivan. As you see in this page:

http://www.neobits.com/do/dtls?pid=9583

This card is a bundle,  wich means supports both, FXO, and FXS. FXO
ports should be used to connect your 3 telco lines, and FXS port to
connect some phones.

Regards


On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote:
 Hi yet again,

 Is a TDM13B card the correct one if I have three phone lines and I
 want to use the extra port to connect a normal phone?

 Thanks!
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[asterisk-users] ISDN / Multiplink PPP (ZapRAS)

2006-09-09 Thread Brent Franks


Hello,

I have found and read Steven Critchfields writeup on how to use ZapRas
(Thanks Critch!), however I am a bit confused.

His write up is here:
http://copilotconsulting.com/mail-archives/asterisk.2003/msg01030.html

Currently we have a full PRI (23B Channels, 1D) coming into our
Asterisk Box using a TE110P.  Voice calls work great, and I have it
all configured up.

Now, I would also like to be able to use this box to establish a 128k
(possibly 256k multilink) PPP connection to an ISP who supports 128k
(2B's) or 256k (4B's) connection.

One thing I don't get understand is, is the Asterisk/Linux box the
modem?  I am confused when he states that he used a Ascend Pipeline
75 as I don't really understand how that fits into the equation.

Can't asterisk dial our end ISP, and then I can set up routing at the
linux level to interface with ppp0 device for example?

Any help would be greatly appreciated!

Thanks so much,

- Brent
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[asterisk-users] DID Provider in Thailand


Does anyone know of a DID provider in Thailand?

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Re: [asterisk-users] MSSQL connection

2006-09-09 Thread Kevin Smith


Thanks Tim,

That was my first thought as well but then I thought, might as well give 
it a try. But it is turning into a hassle more then anything. I already 
have a PHP script wrote to for MySQL so the conversion to MSSQL 
shouldn't be bad.


Thanks,
Kevin



Tim Panton wrote:


On 9 Sep 2006, at 00:42, Kevin Smith wrote:


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?


Also configuration asterisk to use both modules. Any good tips on 
that, I followed the steps provided by the following pages:


http://www.voip-info.org/wiki/view/FreeTDS
http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc

But this is the error I get: (note: some information has been changed 
for security, such as 'user' and pass was changed to phone)


# isql -v MSSQL-astersik phone phone
[S1000][unixODBC][FreeTDS][SQL Server]Unable to connect to data source
[28000][unixODBC][FreeTDS][SQL Server]Login incorrect.
[][unixODBC][FreeTDS][SQL Server]Login failed for user 'phone'.
[][unixODBC][FreeTDS][SQL Server]Cannot open database requested in 
login 'cdr'.

Login fails.
[ISQL]ERROR: Could not SQLConnect

from odbcinst
[MSSQL-FreeTDS]
Description = FreeTDS ODBC driver for MSSQL
Driver  = /usr/lib/libtdsodbc.so
Setup   = /usr/lib/libtdsS.so
FileUsage   = 1

from odbc
[MSSQL-asterisk]
description = Asterisk ODBC for MSSQL
driver  = MSSQL-FreeTDS
server  = XXX.XXX.XXX.XXX
port= 1433
database= cdr
user= phone
password= phone
tds_version = 7.0
language= us_english


Maybe I am just overlooking something or there is something that 
isn't registering with me that is under my nose. Any help would be 
appreciated. My guess is it is an error between the keyboard and 
chair  ;).


We have just been through this - but with Oracle - and came to the 
conclusion that
we didn't want to tightly couple asterisk with the DB, we felt it 
could be a performance
hit - on both sides - plus it meant allowing ODBC traffic over a 
network we couldn't

secure.

In the end we got a script written that reads the Master.csv file, 
turns the new

data into XML and does an HTTP Post of the data to a web service running
on the Oracle system which parses the XML and inserts the records in 
the database.


We plan to run the script every few minutes (from cron).


Tim Panton

www.mexuar.com



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Re: [asterisk-users] Call Forwarding in SIP.conf

I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.

-- Original message -- From: [EMAIL PROTECTED]
Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --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

---BeginMessage---
---BeginMessage---
Check your Dial() string to make sure that you haven't mistyped and put
gafachi-o instead of gafachi-out. Specifiying the full host name will also
work.

As a hint, you can refresh these changes with out restarting your server (and
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.

On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
Tim, this is the way I have Gafachi set up in sip.conf and works well with
channels that have an ATA attached to it but not the virtual one. I have
changed the host in extensions.conf to the .sip.gafachi.com.
But I have calls on the server and cannot restart it yet. I'll keep you
posted and thanks for the feedback.

[gafachi-out]
type=peer
secret=xx
username=x
fromuser=x
fromdomain=xxx
host=.sip.gafachi.com
;usereqphone=yes; This provider requires ;user=phone on
URI ;nat=yes
rtptimeout=60
dtmfmode=rfc2833

-- Original message --
From: Tim St. Pierre [EMAIL PROTECTED]

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--
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]

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

2006-09-09 Thread Gunnar Schaller

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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Re: [asterisk-users] Call Forwarding in SIP.conf

If you don't set the callerID in the channel, it will get passed on as-is.  
Don't change it, and it will stay the same.

-TIm

On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
 I have a follow up question. How do I pass on the caller ID of the call I'm
 forwarding to the other party? I can pass on the channels caller ID but
 prefer to pass on the forwarding party's number instead.

 -- Original message --
 From: [EMAIL PROTECTED]

 Thanks all. It works fine now.

 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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[asterisk-users] Intel Based G.729 and SVN-trunk-r42453

2006-09-09 Thread Jason Lee

Hi,I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at 
http://pastebin.ca/165220Any assistance on this would be appreciated.-- Regards,Jason

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Re: [asterisk-users] Call Forwarding in SIP.conf


I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf

[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed on!host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

---BeginMessage---
If you don't set the callerID in the channel, it will get passed on as-is.  
Don't change it, and it will stay the same.

-TIm

On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
 I have a follow up question. How do I pass on the caller ID of the call I'm
 forwarding to the other party? I can pass on the channels caller ID but
 prefer to pass on the forwarding party's number instead.

 -- Original message --
 From: [EMAIL PROTECTED]

 Thanks all. It works fine now.

 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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RE: [asterisk-users] Call Processing Slow 11 seconds

2006-09-09 Thread G.Jacobsen




In 
case you use an adapter or voip phone: Did you try to press hash # after the 
number ? - thenthe adapter/voip phonedials immediately and doesnt 
wait for the next digit timeout.

Cheers

Gerry


  -Original MessageFrom: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of 
  [EMAIL PROTECTED]Sent: Samstag, 9. September 2006 
  15:15To: asterisk-users@lists.digium.comSubject: 
  [asterisk-users] Call Processing Slow 11 seconds
  I'm having some slowness issue with Asterisk. When a number is dialed it 
  takes 11 seconds before it rings out. I been considering using openser for the 
  call processing and leaving asterisk for voicemail and conference bridge. I 
  get a dialtone rightaway when the receiver is picked up but after dialing the 
  number but within asterisk extensions and pstn numbers takes 11 seconds before 
  ringing out. Anyone else experiencing this. I use Asterisk 
1.2.3
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Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453

2006-09-09 Thread Daniel Pocock




Jason Lee wrote:


Hi,

I was testing the intel based G729 codec on SVN-trunk-r42453 following 
the
new instructions for compiling with SVN trunk and it in preliminary 
tests it

works ok for some calls but I found when one end of the call is an IVR or
Music On Hold the sound gets all distorted and asterisk segfaults. You 
can

view the backtrace at http://pastebin.ca/165220

Any assistance on this would be appreciated.


Have you compiled with debugging symbols instead of CPU optimization?

Can you type `bt' after the segfault, to give us some more detail?

How long into the call does this happen?





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RE: [asterisk-users] Call Processing Slow 11 seconds

Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated.

-- Original message -- From: "G.Jacobsen" [EMAIL PROTECTED]

In case you use an adapter or voip phone: Did you try to press hash # after the number ? - thenthe adapter/voip phonedials immediately and doesnt wait for the next digit timeout.

Cheers

Gerry

-Original MessageFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Samstag, 9. September 2006 15:15To: asterisk-users@lists.digium.comSubject: [asterisk-users] Call Processing Slow 11 seconds
I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3

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Re: [asterisk-users] Call Processing Slow 11 seconds

2006-09-09 Thread Alberto Sagredo


Yes you could script a dialplan putting ... and S0 (zero) at the end.

An example :

(xxS0) It will dial 6 digits directly when you enter the 6th.

You could learn how to adapt your Linksys dialplan looking this wiki.

http://voip.wikispaces.com/

[EMAIL PROTECTED] escribió:
Yes that works. I'm using Linksys adapter, is there a code I can put 
in the dial plan to prevent users from putting # after the number? I 
have a lot of people on the server and cannot ask them all to be 
pushing # after every call. Thanks for the tip and any help will be 
appreciated.
 


-- Original message --
From: G.Jacobsen [EMAIL PROTECTED]
In case you use an adapter or voip phone: Did you try to press
hash # after the number ? - then the adapter/voip phone dials
immediately and doesnt wait for the next digit timeout.
 
Cheers
 
Gerry
 


-Original Message
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
*Sent:* Samstag, 9. September 2006 15:15
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Call Processing Slow 11 seconds

I'm having some slowness issue with Asterisk. When a number is
dialed it takes 11 seconds before it rings out. I been
considering using openser for the call processing and leaving
asterisk for voicemail and conference bridge. I get a dialtone
rightaway when the receiver is picked up but after dialing the
number but within asterisk extensions and pstn numbers takes
11 seconds before ringing out. Anyone else experiencing this.
I use Asterisk 1.2.3




Asunto:
RE: [asterisk-users] Call Processing Slow 11 seconds
De:
G.Jacobsen [EMAIL PROTECTED]
Fecha:
Sat, 9 Sep 2006 17:20:05 +
Para:
Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com


Para:
Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com



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Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453

2006-09-09 Thread Jason Lee

I recompiled with debuging options...both bt and btfull outputs http://pastebin.ca/165250Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash.
I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the whole time I got nothing but distortion.
On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote:
Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it
 works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at 
http://pastebin.ca/165220 Any assistance on this would be appreciated.Have you compiled with debugging symbols instead of CPU optimization?Can you type `bt' after the segfault, to give us some more detail?
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 http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Jason LeeOmegaServ
[EMAIL PROTECTED]Direct Line: (204) 480-1238Toll Free: (866) 664-7786 Ext 200http://www.omegaserv.com
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Re: [asterisk-users] What don't I get about SIP?

2006-09-09 Thread John Marvin


Mike wrote:


Did I misread the Asterisk wiki pages, because I believed that when a
pattern was present, the pattern takes precedence over any real
extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)?


It's the opposite. Asterisk always uses the most specific match for an 
extension, i.e. anything that matches _1XXX will take precedence over 
_, but if it matches _12XX that will take precedence over _1XXX, etc.


John
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Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453

2006-09-09 Thread Daniel Pocock




Jason Lee wrote:


I recompiled with debuging options...

both bt and btfull outputs http://pastebin.ca/165250
Before I recompiled it gave me a second of audio then I got nothing but
distortion for 5 seconds then asterisk would crash.
I retested after compiling it with just a call between two local 
devices one
using ulaw and the other using g729 and I'm getting nothing but 
distortion.
I then tried calling music on hold and it took 3 minutes to crash the 
whole

time I got nothing but distortion.


This suggests that someone/something gave the command `stop now'

Can you send the backtrace from a segfault?



On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote:





Jason Lee wrote:

 Hi,

 I was testing the intel based G729 codec on SVN-trunk-r42453 following
 the
 new instructions for compiling with SVN trunk and it in preliminary
 tests it
 works ok for some calls but I found when one end of the call is an IVR
or
 Music On Hold the sound gets all distorted and asterisk segfaults. You
 can
 view the backtrace at http://pastebin.ca/165220

 Any assistance on this would be appreciated.

Have you compiled with debugging symbols instead of CPU optimization?

Can you type `bt' after the segfault, to give us some more detail?

How long into the call does this happen?


 



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Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453

2006-09-09 Thread Jason Lee

Sorry about that. I thought I had the right core dump. I retried again and the output from bt and bt full is at http://pastebin.ca/165289It took 1min 50seconds of nothing but distortion before asterisk segfaulted
-- Regards,JasonOn 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote:
Jason Lee wrote: I recompiled with debuging options...
 both bt and btfull outputs http://pastebin.ca/165250 Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash.
 I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the
 whole time I got nothing but distortion.This suggests that someone/something gave the command `stop now'Can you send the backtrace from a segfault? On 9/9/06, Daniel Pocock 
[EMAIL PROTECTED] wrote: Jason Lee wrote:  Hi,   I was testing the intel based G729 codec on SVN-trunk-r42453 following
  the  new instructions for compiling with SVN trunk and it in preliminary  tests it  works ok for some calls but I found when one end of the call is an IVR
 or  Music On Hold the sound gets all distorted and asterisk segfaults. You  can  view the backtrace at http://pastebin.ca/165220
   Any assistance on this would be appreciated.  Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail?
 How long into the call does this happen?   ___
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RE: [asterisk-users] What don't I get about SIP?

2006-09-09 Thread Mike

It certainly makes sense, and I tried it...it works, you are right.

So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting 

Mike

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 John Marvin
 Sent: September 9, 2006 2:05 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What don't I get about SIP?
 
 Mike wrote:
 
  Did I misread the Asterisk wiki pages, because I believed 
 that when a 
  pattern was present, the pattern takes precedence over any real
  extensions? (i.e. if I have both 1234 and _1XXX as 
 extensions in a context)?
 
 It's the opposite. Asterisk always uses the most specific 
 match for an extension, i.e. anything that matches _1XXX will 
 take precedence over _, but if it matches _12XX that will 
 take precedence over _1XXX, etc.
 
 John
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Re: [asterisk-users] Another (quick) Polycom 501 question

2006-09-09 Thread Kevin Smith


Hi Mike,

As far as I know, you need to at least start the dialing (ie New call, 
speaker, etc) for the digitmap to even come into play.


The only settings that I am aware of that you can try to change are 
dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf.


Kevin

Mike wrote:

Hi all,
 
That's my last one for a while (I hope).
 
How can I (if at all possible) make the 501 turn on the speaker phone 
as soon as a digit is dialed (if the handset is not lifted)? Sort of 
like what a normal speakerphone does.
 
The reason I want this is I want the 501 digitmap to be taken into 
consideration even if the handset isnt lifted and the speakerphone 
button isn't consciously pressed.  For all those users who don't want 
to press send, but like dialing without lifting the handset (and can't 
be bothered to press the speakerphone button).  Yes I know it's 
capricious, but we have the users we have...
 
Yes, I have read the admin manual, but couldn't find the info.  I am 
assuming I just don't know what to look for, but that this 
functionality exists.
 
 
 
Mike



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[asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54

2006-09-09 Thread FRANCISCO PEREZ-LANDAETA



hi i need helpl configuring  a quintum tenor analog gateway using sip with 
asterisk.

anyone,
help is appreciated
the model of the gteway is asm200 i need the settings to configure it with 
asterisk.
for some reason it registers with asterisk but when try to call the 
extension from the quintum it is not recognized.

help help help

thanks


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Today's Topics:

   1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED])
   2. RE: Call Processing Slow 11 seconds (G.Jacobsen)
   3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock)
   4. RE: Call Processing Slow 11 seconds ([EMAIL PROTECTED])
   5. Re: Call Processing Slow 11 seconds (Alberto Sagredo)
   6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee)
   7. Re: What don't I get about SIP? (John Marvin)
   8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock)
   9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee)
  10. RE: What don't I get about SIP? (Mike)


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Message: 1
Date: Sat, 09 Sep 2006 17:12:54 +
From: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Call Forwarding in SIP.conf
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
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Message: 2
Date: Sat, 9 Sep 2006 19:17:23 +0300
From: G.Jacobsen [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Call Processing Slow 11 seconds
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
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In case you use an adapter or voip phone: Did you try to press hash # after
the number ? - then the adapter/voip phone dials immediately and doesnt 
wait

for the next digit timeout.

Cheers

Gerry

  -Original Message
  From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
  Sent: Samstag, 9. September 2006 15:15
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Call Processing Slow 11 seconds


  I'm having some slowness issue with Asterisk. When a number is dialed it
takes 11 seconds before it rings out. I been considering using openser for
the call processing and leaving asterisk for voicemail and conference
bridge. I get a dialtone rightaway when the receiver is picked up but after
dialing the number but within asterisk extensions and pstn numbers takes 11
seconds before ringing out. Anyone else experiencing this. I use Asterisk
1.2.3
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Message: 3
Date: Sat, 09 Sep 2006 18:23:37 +0100
From

Re: [asterisk-users] Call Processing Slow 11 seconds

Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing.

-- Original message -- From: Alberto Sagredo [EMAIL PROTECTED] Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. --
O
riginal message -- From: "G.Jacobsen" <[EMAIL PROTECTED]> In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been conside
ring u
sing openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: "G.Jacobsen" <[EMAIL PROTECTED]> Fecha: Sat, 9 Sep 2006 17:20:05 + Para: "Asterisk Users Mailing List - Non-Commercial Discussion" Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Samy Antoun

--- Bill Maidment [EMAIL PROTECTED] wrote:

 Hi
 I've just tried to compile the zaptel-1.2.9 release and I get the
 following error:


Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when
compiling zap:

make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found
make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found
make[3]: *** No rule to make target
`/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed
by `/usr/src/zaptel/wct4xxp/vpm450m.o'.  Stop.
make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2
make[1]: *** [_module_/usr/src/zaptel] Error 2
make: *** [linux26] Error 2

Hope someone has a workaround for this problem


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[asterisk-users] Problems configuring Polycom 301

2006-09-09 Thread Jim Freeze

HiI have successfully been running with several Polycom SoundPoint 501phones and recently purchased some Polycom 301 phones.However, I can't seem to get the phones to register. The phone seesthe asterisk server, but all calls our are busy.
The only difference for 'sip show peer xxx' for a working 501 phone anda non working 301 phone is:asterisk1*CLIAddr-IP : 
192.168.80.204 Port 5060 # 501Addr-IP : (Unspecified) Port 0 # 301
'sip show peers' returns:asterisk1*CLI sip show peers
Name/usernameHostDyn Nat ACL Port Status720/720(Unspecified)D0UNKNOWN
712/712192.168.8.205 D5060 OK (80 ms)
711/711192.168.8.203 D5060 OK (84 ms)
710/710192.168.8.204 D5060 OK (98 ms)Any 301 configuration tips would be appreciated.Thanks-- Jim Freeze
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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Nigel Godfrey


The work around is at:
http://www.sineapps.com/news.php?rssid=1496

On 09/09/06, Samy Antoun [EMAIL PROTECTED] wrote:

--- Bill Maidment [EMAIL PROTECTED] wrote:

 Hi
 I've just tried to compile the zaptel-1.2.9 release and I get the
 following error:


Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when
compiling zap:

make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found
make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found
make[3]: *** No rule to make target
`/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed
by `/usr/src/zaptel/wct4xxp/vpm450m.o'.  Stop.
make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2
make[1]: *** [_module_/usr/src/zaptel] Error 2
make: *** [linux26] Error 2

Hope someone has a workaround for this problem


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[asterisk-users] ztdummy installed but choppy audio warning on load

2006-09-09 Thread Nigel Godfrey


On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium hardware.

modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives:
[EMAIL PROTECTED] ~]# lsmod
Module  Size  Used by
ztdummy 3924  0
zaptel206852  5 ztdummy

When asterisk starts it logs warnings:

Sep  9 20:28:00 WARNING[2645] res_musiconhold.c: Unable to open pseudo
channel for timing...  Sound may be choppy.
Sep  9 20:28:02 WARNING[2645] chan_iax2.c: Unable to open IAX timing
interface: No such file or directory

I've Googled the error message, but to no avail. Any thoughts, please?


nigel.
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Re: [asterisk-users] ztdummy installed but choppy audio warning on load

2006-09-09 Thread Daniel Pocock



zap show status

will tell you if Asterisk is really using ztdummy

Make sure you have chan_zap.so enabled in modules.conf (or that it isn't 
disabled with a noload declaration)



Nigel Godfrey wrote:


On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium hardware.

modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod 
gives:

[EMAIL PROTECTED] ~]# lsmod
Module  Size  Used by
ztdummy 3924  0
zaptel206852  5 ztdummy

When asterisk starts it logs warnings:

Sep  9 20:28:00 WARNING[2645] res_musiconhold.c: Unable to open pseudo
channel for timing...  Sound may be choppy.
Sep  9 20:28:02 WARNING[2645] chan_iax2.c: Unable to open IAX timing
interface: No such file or directory

I've Googled the error message, but to no avail. Any thoughts, please?


nigel.
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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Samy Antoun

--- Nigel Godfrey [EMAIL PROTECTED] wrote:
 The work around is at:
 http://www.sineapps.com/news.php?rssid=1496

Thanks, I'll give it a try.


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[asterisk-users] Scope of contexts

2006-09-09 Thread Rene

Hi all,

I am trying to understand contexts a bit better. The problem I have is
when you know when a context is finished. Is this when a new context
starts?

Example:
[context1]
exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

[context2]
exten = 6394,1,Dial(Local/6275/n)

When your call starts at context1, will it automatically go to context2
when context1 is finished?

Hope someone can shed some light on this

Thanks,
Rene

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Re: [asterisk-users] What don't I get about SIP?

2006-09-09 Thread John Marvin


Mike wrote:

It certainly makes sense, and I tried it...it works, you are right.

So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting 



Interesting. I got my information from Asterisk: The Future of 
Telephony (in the dialplan chapter). Perhaps the wiki page refers to 
1.0 behaviour, and 1.2 behaviour is what is defined in Asterisk: TFOT?


My experimentation so far has shown the Asterisk: TFOT information to be 
correct. I haven't played around with #include, which the wiki says can 
change the dialplan extension sorting. I may have to experiment with 
that to see if it has any effect.


I hope that what I said is correct regardless, because it makes the most 
sense and is less likely to cause weird issues when changing the order 
of #includes, etc.


John
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Re: [asterisk-users] Scope of contexts

2006-09-09 Thread Doug Lytle


Rene wrote:

Hi all,

I am trying to understand contexts a bit better. The problem I have is
  



This should help:

http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/c650.html


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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Re: [asterisk-users] Scope of contexts

2006-09-09 Thread Moises Silva


Yep, that should help, and the short answer to you question is NO.

Regards

On 9/9/06, Doug Lytle [EMAIL PROTECTED] wrote:

Rene wrote:
 Hi all,

 I am trying to understand contexts a bit better. The problem I have is



This should help:

http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/c650.html


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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Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines

Thanks Daniel, your advice helped. It was the call waiting not working, which I thoughtwas working because I had selected 'No' beside 'Disable Call-Waiting'. But for call waiting to work properly, I also needed to select 'No' beside '
Enable Call Features' and then dial *70 to enable it on Asterisk.Now it works perfect.
On 9/8/06, Daniel Salama [EMAIL PROTECTED] wrote:

You need to enable call waiting on the phone's config.

- Daniel

On Sep 8, 2006, at 9:35 AM, Zeeshan Zakaria wrote:

First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, all lines are busy, please leave your message after the tone. I tried resetting phone to factory default setting too, but still it does the same. Same extension if configured on X-TEN, it works with no problems for all available free lines. Grandstream phone should go upto 11 lines and only for 12th call should say that lines are busy.

What I need to configure in this Grandstream phone which I haven't figured out yet. I've firmware 1.1.0.16
-- Zeeshan A Zakaria

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Re: [asterisk-users] How to send correct Caller ID on PRI

I figured out how to send the caller ID, it is working. But now I am trying to send the Company Name along with caller ID, and that is not working. What could be the reason for that. I am using SerCallerID. Is there something else I should use to send the alphabets?
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Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging

I am still trying to make it work. Where did you get firmware version 1.1.1.9. On there website they have only 1.1.0.16
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Re: [asterisk-users] Grandstream, how to use the configuration tool

I'll try to use it, but in future I think I'll get some other phones with better configuration utilities.
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Re: [asterisk-users] Roundrobin not working on PRI

I sent the incoming calls to different queues, and now everything is working fine. Just changed the MoH option of the first queue to ring tone, so that the callers hears the ring going, otherwise they'll feel uncomfortable why music has started all of a sudden.
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[asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.

In my queues, I wanted callers to listen dial tone going, instead of listening the music, to I used option 'r' in queue command. Now it doesn't announce the position of caller and estimated hold time etc. Is this normal for this setting or I am soing something wrong. If I don't use option r and leave queue as it is, 
i.e. playing MoH, then all announcements work ok.-- Zeeshan A Zakaria 
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[asterisk-users] Whcih phones are better for mass deployment

I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use.


For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones?
-- Zeeshan A Zakaria 
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Re: [asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.

2006-09-09 Thread Michiel van Baak

On 18:47, Sat 09 Sep 06, Zeeshan Zakaria wrote:
 In my queues, I wanted callers to listen dial tone going, instead of
 listening the music, to I used option 'r' in queue command. Now it doesn't
 announce the position of caller and estimated hold time etc. Is this normal
 for this setting or I am soing something wrong. If I don't use option r and
 leave queue as it is, i.e. playing MoH, then all announcements work ok.

I noticed the same thing couple of months ago.
Talked about it here and on irc but noone could give me a
clear answer.
I'm now still stuck with moh because I really want the
announcements to be played :(

If someone has an mp3 or wav of ringing, please post it...

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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

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


Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.

-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] 
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. 

For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Zeeshan A Zakaria 

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Re: [asterisk-users] Call Processing Slow 11 seconds

2006-09-09 Thread Andres


[EMAIL PROTECTED] wrote:

Thanks, I tried that and did not work for me. My users are calling US 
number and without the # at the end of the last digit dials it takes 
11 seconds before it starts ringing.
 


If you are dialing 11 digits then set the Linksys Dial Plan to: 
(1xx), the phone will dial out right after the last digit.  You 
will need to modify the dial plan if there are other patterns as well 
that are expected.


--
Andres
Technical Support
http://www.telesip.net

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Re: [asterisk-users] How to send correct Caller ID on PRI

2006-09-09 Thread Doug Lytle


Zeeshan Zakaria wrote:
I figured out how to send the caller ID, it is working. But now I am 
trying to send the Company Name along with caller ID, and that is not 
working. What could be the reason for that. I am using SerCallerID. Is 
there something else I should use to send the


Most phone companies will not allow for you to set callerid name, only 
number.


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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Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-09 Thread Chris Dos

Before the phone starts it boot process, I edit the phone settings under server
information, enter the IP of the sip server and also the protocol to use, UDP 
Only.

Chris

David Gagnon wrote:
 Hi Chris,
 
   I'm would like to get more information about this problem. Why the
 phone isn't registering properly with the new firmware and what did you mean
 by hard coding the sip settings?
 
 Thx
 
 David
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Chris Dos
 Envoyé : 7 septembre 2006 15:30
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] Polycom new firmware and bootrom
 
 Not to mention the feature that the new firmware and bootrom that prevent
 it
 from registering with the Asterisk server unless you hard code the sip
 settings.
 
   Chris
 
 Jessee J Holmes wrote:
 Also keep in mind that as of right now, the latest bootrom and firmware
 available from Polycom (and thus your reseller) are Bootrom 3.2.2 and
 Firmware 2.0.1

 The 2.0.1 firmware is new as of a day or two and include some
 enhancements for buddy lists and shared presence as well as newly added
 secured TLS support (according to Polycom).



 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 7, 2006, at 1:19 PM, Douglas Garstang wrote:

 That process is worse than pulling teeth!

 -Original Message-
 *From:* Jessee J Holmes [mailto:[EMAIL PROTECTED]
 *Sent:* Thursday, September 07, 2006 11:25 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Polycom new firmware and bootrom

 All authorized Polycom resellers will have access to this firmware
 and are required to provide this firmware to you. Contact the
 reseller you purchased the Polycom phone from.


 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 7, 2006, at 11:55 AM, Bruce Reeves wrote:

 Typically you have to go to a reseller who you purchased Polycom
 equipment from. Even then it can be tricky since they have to
 find away to get you the files with out upsetting Polycom.

 On 9/7/06, *Douglas Garstang* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 Polycom are analy retentive about giving out software updates.

  -Original Message-
  From: Nathan Alberti [mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]]
  Sent: Thursday, September 07, 2006 10:25 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Polycom new firmware and bootrom
 
 
 
 
  Stupid question where did you find it ?
 
 
  Looked at their site downloads and under the extranet site
 but could
  only see old versions.
 
  Nathan.
 
  On 07/09/2006, at 10:21 AM, Chris Dos wrote:
 
   Well, it seems that Polycom has release new firmware
 2.0.1 and
   bootrom 3.2.2.
   I've proceded to upgrade all my ip430 phones because they
 were
   essentially
   broken with the original firmware.
  
   All the phones boot up fine now, grab their files.  They
  just won't
   talk to the
   asterisk server any more.   I just figured out that I
 need to hard
   code the sip
   server and tell it to talk udp only.  After this, the
  phones worked
   again.
  
   Any idea on what I need to configure to fix the phones so
  they will
   know which
   server to talk to and only talk to it via udp?
  
   Chris
  
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Re: [asterisk-users] Whcih phones are better for mass deployment

2006-09-09 Thread Michael Graves



Polycom &  Aastra are both great in this manner.



Michael



--Original Message Text---

From: [EMAIL PROTECTED]

Date: Sun, 10 Sep 2006 00:18:37 +



Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.

 

-- Original message -- 

From: "Zeeshan Zakaria" [EMAIL PROTECTED] 

I am having hard time with grandstream phones for a 30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. 

 

For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? 

-- 

Zeeshan A Zakaria 



--NextPart_Webmail_9m3u9jl4l_7096_1157847517_1-- 






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[asterisk-users] What really happens between Asterisk and an SPA-3000?

2006-09-09 Thread Larry Alkoff

I'm trying to get a clear understanding just how calls are routed in a 
mixed SPA3k and Asterisk system.


This is my present (incomplete) understand and I'd appreciate any 
corrections.  I'm especially interested in what happens between Asterisk 
and an SPA3k.


Note:
-
POTSaudio refers to POTS line audio signal.
SIPaudio refers to sip IP packets.
User is us with the Sipura and Asterisk goodies.
Caller is the outside person calling us.

No power to SPA3k:
--
failover mode.
FXO line (CO line) audio is connected directly to FXS line (POTS phones) 
by a relay.


Incoming SIP call to Asterisk:
--
Handled by Asterisk as SIP to SIP. Call never touches SPA3k.

Incoming POTS call to SPA3k:

POTSaudio converted to SIPaudio signal in SPA3k.
SIPaudio forwarded to PSTN-IN extension on Asterisk server (ext 201 for me).
Asterisk should ring both SIP phones and POTS phones
using context in extensions.conf so User can pickup either.

If POTS phone picked up:

Does POTSaudio go directly back and forth over POTS line
or is there a SIP conversion anywhere?

If SIP phone picked up:
---
SIPaudio from SPA3k Caller is heard by User
SIPaudio from User goes out over PSTN-OUT to SPA3k
which converts SIPaudio to POTSaudio and out Line to Caller.

Outgoing calls:
---

If 7 digit local or 911, outgoing context in extensions.conf
routes call to SPA3k and out PSTN-SPA3k gateway.
SIPaudio to SPA3k which converts it to POTSaudio.

Other calls are routed either to SIP extensions
or SIP provider. SPA3k is out of the picture.

Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Whcih phones are better for mass deployment

Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free?
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Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging

I finally made the paging to work. But the only thing which I had to change was the number to dial. As in the instructions, it is _**1, but it didn't work for me and I used 333 instead. All other settings are the same. Now the reception phone's one Speed Dial key is assigned ext 333, pressing which activates speaker phones of all the phones in the office for paging.
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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Bill Maidment


Nigel Godfrey wrote:

The work around is at:
http://www.sineapps.com/news.php?rssid=1496


Thank you. I'd forgotten about SVN. Works like a charm.
Cheers
Bill

--
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Maidment Enterprises Pty Ltd
www.maidment.com.au

si hoc non legere potes tu asinus es

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


for mass deployment the Linksys will allow you to update your routers with a tftp server.. You can have the routers always download their software from the tftp server, that way you have the latest on the server for upgrade software. The reason that I don't like granstream is their bad customer support, they live you on your own basically.

-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? 

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Re: [asterisk-users] Scope of contexts

Nope.  A context ends when a new one starts.  The only way for a call to 
continue is to have a maching extension, and the next higher priority.  If 
you want a call to continue in another context, you need to use the Goto() 
application. 

On September 9, 2006 19:04, Rene wrote:
 Hi all,

 I am trying to understand contexts a bit better. The problem I have is
 when you know when a context is finished. Is this when a new context
 starts?

 Example:
 [context1]
 exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

 [context2]
 exten = 6394,1,Dial(Local/6275/n)

 When your call starts at context1, will it automatically go to context2
 when context1 is finished?

 Hope someone can shed some light on this

 Thanks,
 Rene

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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

I am a really big fan of Aastra phones.

It's a splinter company of Northern Telecom, so their quality is very good.

Provisioning is done via a text file on either a tftp or an ftp server.  There 
is a global file and a per phone file.  When you have a good set of config 
files built, you can include an option for the phones to check the files 
every day at a predetermined time and reboot if there are any changes.  You 
can also send an SIP NOTIFY to cause the phones to update their config if you 
change something and need it applied immediately.  There is no config utility 
needed, as the files are human readable.  There is an encryption utility if 
you are concerned about security.  

When you deploy a new phone, you need only set the TFTP server address.  After 
that, the phone can get all it's settings from the server.  I have about 40 
of them deployed at client sites that I usually don't have access to.  I can 
change everything from here.  

Sound quality is great, most of them support PoE and have a passthru ethernet 
port.  The displays are backlit, there is a full duplex speaker phone and 
headset jack on all models.  There is also a built in directory function that 
loads from a .csv file on the server.  BLF support is good on the 9133i and 
the 480i.  

I can't say enough good things about these phones.  Manufacturer support is 
also very good.  Free firmware downloads from the website and good 
documentation.

-Tim

On September 9, 2006 18:51, Zeeshan Zakaria wrote:
 I am having hard time with grandstream phones for a 30 phone setup. When a
 change in configuration is required, I have to change their configurations
 manually for almost all of them. Their configuration utility is not very
 straight forward to use.

 For my next installation, I would prefer some other phones with better
 configuration and remote accress utility. My question to those of you with
 more experience, what IP phones are better for mass deployment and easy
 management of updates and configurations? Or what other solution is better
 for mass deployment of phones?

-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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

I'll speak on the Aastra, since that is what I know, although most of this 
applies to Polycom as well.

There is no windows software needed at all.  Personally, I haven't been a 
Microsoft customer in more than half a decade.  Their operating systems are 
not appropriate for telecom applications.  You can do all your configuration 
with a text editor.  This is good for several reasons:

1) You can administer the phone config directly on the server over an ssh 
connection.

2) A shell script can create phone config files.  I have a shell script that 
appends to extensions.conf, sip.conf, voicemail.conf, and creates a phone 
config file.  You can automate things very easily this way.

3) Since there is only one setting to put into the phone to use the remote 
config, a customer can be talked through a factory reset and reset the server 
address over the phone if they really screw things up.  You can have an 
inventory of phones with your config server address already set.  All you 
need is the phone MAC address, and you can build a config file.  This means 
that you could send phones to customers without them having decided what to 
do with them yet.  

4) These phones are reliable and well constructed.  They will require less 
maintenance, and will last longer.

5) They have features that are appropriate to a business environment.

6) They will usually find their way around a NAT firewall, so they are 
essential plug-and-play at the customer site.

Let me know if you have any more questions.

-Tim


On September 9, 2006 23:01, Zeeshan Zakaria wrote:
 Can you explain a little bit what make them better for mass deployment. Do
 they have Windows based software to communicate with all the installed
 phones and upgrade them and also to remotely monitor them. Is there a
 separate cost for these software tools or are they free?

-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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

Thanks for the info. In my next installation, I'll try those phones.
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Re: [asterisk-users] Whcih phones are better for mass deployment

2006-09-09 Thread Andrew Kohlsmith

On Saturday 09 September 2006 18:51, Zeeshan Zakaria wrote:
 For my next installation, I would prefer some other phones with better
 configuration and remote accress utility. My question to those of you with
 more experience, what IP phones are better for mass deployment and easy
 management of updates and configurations? Or what other solution is better
 for mass deployment of phones?

Polycom.  Nothing I've ever seen or used works better for mass deployment.

-A.
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[asterisk-users] Grandstream GX-2000 Remote Login Problem

On my grandstream phones, I am having problem that I can't login in remotely. With in the local network, they work ok, but outside the network, they don't. All the NAT settings are ok and I can log into the Asterisk server from X-Ten, no problem. But Grandstream phone doesn't login. I've port forwardedSIP 5060-5080, RTP 1-14000. In 
rtp.conf, ports range from 1-14000.

Same extensions when configured on x-ten, with RTP listening on 1, they login with no problem. But when configured on grandstream, they don't configure. What am I missing here.-- Zeeshan A Zakaria 

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Re: [asterisk-users] Grandstream GX-2000 Remote Login Problem

2006-09-09 Thread William Piper

I've noticed that Grandstream works better using stun and not port forwarding your router.

Try setting stun.xten.com or stun.fwdnet.net in your GS2000 and make sure sip.conf has nat=yes. It should work fine.

Also, i've noticed that Linksys wireless with speed booster has something in it that is blocking VoIP. I set DMZ, done port forwarding... nothing... I could get it to register but no return sound... it was being blocked. Just something to keep in the back of your mind.


bp
On 9/10/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:


On my grandstream phones, I am having problem that I can't login in remotely. With in the local network, they work ok, but outside the network, they don't. All the NAT settings are ok and I can log into the Asterisk server from X-Ten, no problem. But Grandstream phone doesn't login. I've port forwardedSIP 5060-5080, RTP 1-14000. In 
rtp.conf, ports range from 1-14000.

Same extensions when configured on x-ten, with RTP listening on 1, they login with no problem. But when configured on grandstream, they don't configure. What am I missing here.-- Zeeshan A Zakaria 
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[asterisk-users] Streaming audio for MoH