[Asterisk-Users] Snom 360 with Firmware 6.1?

[Koopmann, Jan-Peter]

Hi,

Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I
am currently using Firmware 5.5 without serious problems but wanted to make
sure 6.X will work as well (including subscription etc.)

Kind regards,
  JP


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RE: [Asterisk-Users] Out of Office Auto Reply:

[Koopmann, Jan-Peter]

On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote:

 We use MS Exchange too and, as far as I am aware, it is cognizant of
 mailing list headers and doesn't send OOO notices to mailing list
 postings. The only mailing list from which I receive my own OOO
 notices is one that doesn't have the proper mailing list headers set.


No. Exchange does not honour Precedence headers. It has some funky way of 
determining what is a mailing list and what is not. It does not work very well 
and it has (or had) to be enabled via a registry key. If you don't do this, 
even Exchange 2003 will reply to some mailing lists. But it should not send 
this to every mail but only once day...


Kind regards,
  JP

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Re: [Asterisk-Users] show queue ... Invalid

[Kevin P. Fleming]

- Denis Shaposhnikov [EMAIL PROTECTED] wrote:
 I've found the problem. That's because I've loaded app_queue.so
 before
 chan_sip.so in modules.conf.

That makes perfect sense. Thanks for following up!

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] sip to h323 ... direct RTP?

[Kevin P. Fleming]

- Jeremy McNamara [EMAIL PROTECTED] wrote:
 The problem is 're-inviting' in H.323-jive is very much a non-trivial
 task.

Ahh, OK, then this is a protocol limitation more than an implementation issue. 
Never mind :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Re: fail to make call

[Kevin P. Fleming]

- unplug [EMAIL PROTECTED] wrote:
 Is it the way for asterisk realtime system?
 register:
 UA1 --register-- asterisk1 - store user information in DB
 UA2 --register-- asterisk2  store user information in DB
 UA1 --invite-UA2--- asterisk1  asterisk1 query UA2
 information in DB
-asterisk1 -invite-- UA2
 finally: UA1 ---asterisk1 --UA2  (2 legs keeping in
 asterisk1)

No, this is not supported at this time. If the UAs are behind NATs, then this 
wouldn't work even if the database was set up properly, because UA2's NAT is 
not prepared to accept SIP signaling from asterisk1 anyway.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Re: fail to make call

[Kevin P. Fleming]

- unplug [EMAIL PROTECTED] wrote:
 BTW, do you mean this function will be included in next release? 
 When
 will be the next release available?

No, it will not be in the next release (which is Asterisk 1.4). It may be in 
Asterisk 1.6, scheduled for January or so of 2007, but as I said in my other 
reply, even with the database support set up properly, there are still 
significant problems with operating this way, depending on how the network is 
structured.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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RE: [Asterisk-Users] Snom 360 with Firmware 6.1?

[Mimmus]

Just installed!
Use 6.1.1 (beta) because 6.1 has a few of registration problems.

Bye


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Koopmann, Jan-Peter
 Sent: Friday, June 23, 2006 9:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Snom 360 with Firmware 6.1?
 
 Hi,
 
 Has anybody experience with Snom360 and Firmware 6.X with 
 Asterisk 1.2.X? I am currently using Firmware 5.5 without 
 serious problems but wanted to make sure 6.X will work as 
 well (including subscription etc.)

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Re: [Asterisk-Users] 1.2.9.1 crashed today

[Kevin P. Fleming]

- Steve Totaro [EMAIL PROTECTED] wrote:
 My system never stops processing calls but it slows.  On the CLI, it
 may 
 take five minutes for a command to execute but it does finally execute
 
 (such as sip show peers or show channels)  And calls to the agents sit
 
 in queue for a while before being delivered to an agent even thought
 he 
 agent is available.

There is a problem that has been identified that was caused by a recent change 
in Asterisk, that may be the reason you are seeing these symptoms. It would 
appear to only affect people using chan_agent, and then only if the agents are 
transferring calls they receive.

I am still working on a proper solution for this issue, but hope to get it 
resolved in the next day or so. Stay tuned and watch for a new release :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.

[Kevin P. Fleming]

- Dmytro Mishchenko [EMAIL PROTECTED] wrote:
 In this case during all conversation SIP packets contains 
 Call-ID: [EMAIL PROTECTED] 
 but the final BYE packet from adapter contains 
 Call-ID: [EMAIL PROTECTED]
 Is such scenario correct from SIP protocol point of view?

No, it is not valid. The Call-ID is used to uniquely identify the SIP dialog, 
and must remain the same.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!

[Kevin P. Fleming]

- Matt [EMAIL PROTECTED] wrote:
 Is there something volume wise different about the internal mp3 and
 wav players?

Probably. People usually use a volume factor of somewhere between .25 and .40 
in sox when preparing files for MOH usage.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Dave Cotton]

On Thu, 2006-06-22 at 20:53 -0400, Steve Totaro wrote:

  From their README
 
 Disclaimer -- The information provided by CDRTool documentation 
 is not always enough to successfully complete the installation and 
 deployment of CDRTool. Most of the configuration tasks are related to 
 setting up components outside CDRTool environment. Configuration of 
 components like MySQL server, FreeRadius server, Cisco gateways, 
 MediaProxy or SIP Express Router require a project management approach 
 in place of a step by step installation procedure.

This says everything about the OP, looking at the often totally stupid
posts on his part.

C'est la vie.

-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] billing

[Khaled Chehab]

I am using
trixbox ,please any ont knows how to confiure billing on it,

I want to
make a billing ,I created an account at http://x.X.X.X/a2billing but it does work 








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Re: [Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?

[Rostislav Bagrov]

I have the same problem here. There is an function called TIMEOUT and in 
1.2.9.1 it has to be set for specific channel this way: 
Set(TIMEOUT(response)=seconds)
This works on any other but Zap channels. It seem that the default timeout is 
still back there discarding my custom one. My playground is like:

2n GSM Gate - Zap - Asterisk (IVR with phonebook)


I would also apreciate some light on this case.


Thanks.




- Original Message -
From: Benoit Panizzon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu, 22 Jun 2006 23:18:48 +0300
Subject: [Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?


 Hi all
 
 There seam to be a very short timeout waiting for digits being dialed.
 (about 
 6 seconds).
 
 Is there a way to increase that time? I have a phone with integrated address
 
 book and my fingers are just not fast enough to open the menue, select an 
 entry and hit 'dial'.
 
 -Benoit-
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Re: [Asterisk-Users] PRI Issue - Calls being rejected with unacceptable channel

[Dinesh Nair]

On 06/23/06 01:22 Andy Brezinsky said the following:

 Protocol Discriminator: Q.931 (8)  len=47
 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
 Message type: SETUP (5)



  Protocol Discriminator: Q.931 (8)  len=10
  Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
  Message type: CALL PROCEEDING (2)



  Protocol Discriminator: Q.931 (8)  len=14
  Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
  Message type: CONNECT (7)


i may be way offbase with this, but on our PRI calls, we usually have 
asterisk sending an ALERTING between the CALL PROCEEDING and CONNECT. this 
seems borne out by...




 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
 Message type: RELEASE (77)



 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
 Message type: STATUS (125)
 [08 03 83 e5 07]
 Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Transit network (3)
  Ext: 1  Cause: Message not compatible with call state 
(101), class = Protocol Error (6) ]


...Message not compatible with call state STATUS returned by the other 
side. you may want to experiment with a Wait(2) before the Answer().


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
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Re: [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering

[Daniel Salama]

I had the same problem some time ago. Make sure call waiting is NOT  
disabled. This will make the phone receive more calls on the other  
lines.


- Daniel

On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne  
wrote:


I have a network of GXP 2000 phones and would like to know if there  
is a
way to configure the phones so that if there is one person talking,  
and

another call comes in then they can hold/hangup that call and take the
incoming call.

At the moment, when a call comes in and the phone is offhook, then  
that

phone is completely unavailable for that ring session, any call coming
in after that call will of course ring.

Is this limited to the GXP series or does the SNOM phones fix this,  
etc.


Any advice is appreciated of course.
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[Asterisk-Users] GXP-2000 and Shared Line Appearances

[Daniel Salama]

I have a client with 20 GXP-2000s. Everything seems to be working  
fine. However, after a couple of weeks of use, the client is having a  
hard time adjusting to the new IP based phone systems and only misses  
one feature from their old Lucent system.


That is, they had 8 analog lines before and all their old Lucent  
phones showed a button for each line. So, it was easy for anyone to  
say, pick up line 2 or anyone to see which lines were in use.


Is it possible to use the GXP-2000 line buttons or extension buttons  
to show the lines in use, shared by all phones. Since the client is  
purchasing 8 virtual lines, I have them restricted in a call group  
and also with incoming and outgoing call limits. Is it possible for  
all the GXP-2000s to show that line 1 is in use, and so on?


Thanks,
Daniel

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RE : Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[hgaillac-sip]

Steve,

I do hope ag-projects to keep their products with
commercial licences !

Harry

PS:

most of billing (or not) open source projects provide
a good documentation .
  


--- Steve Totaro [EMAIL PROTECTED] a
écrit :

 Alex Robar wrote:
  Harry,
 
  Fisrt ag-projects talk about is product like a
 gpl
  software however they don't provide at least
 some
  documentation for non commercial users .
 
 
  Are they required to release docs? It's a nice
 bonus, but there's 
  nothing forcing them to. If you want docs, you
 could always be helpful 
  and write some yourself.
 
  try to call them !!
  i'll offer you some money .
 
  You can not Call them for some advices ... 
 
 
  And why should you be able to? Again, it's nice to
 have phone support, 
  but in no way is it required. ViciDial doesn't
 have phone support, but 
  people use it. FreePBX doesn't offer phone
 support, but it's pretty 
  popular.
 
  It's really a bad product don't waste your
 time to
  setup it.
  this enterprise must be  fogotten it's
 ag-projects .
  it's not a reliable society ... more and more
 
 
  Sounds to me like complaints from someone who
 couldn't figure out how 
  to get the product setup to me.
 
  If you have valid complaints then by all means,
 post it to the list as 
  a warning... But even then, there's no need to
 openly bash a product 
  or company you don't like. This isn't the forum
 for such things.
 
  Alex
 
  -- 
  Alex Robar
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
  From their README
 
 Disclaimer -- The information provided by
 CDRTool documentation 
 is not always enough to successfully complete the
 installation and 
 deployment of CDRTool. Most of the configuration
 tasks are related to 
 setting up components outside CDRTool environment.
 Configuration of 
 components like MySQL server, FreeRadius server,
 Cisco gateways, 
 MediaProxy or SIP Express Router require a project
 management approach 
 in place of a step by step installation procedure.
 
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[Asterisk-Users] Passing DID to external number?

[Brian McCarey]

Hi,

We run a small 
switchboard using Asterisk and Free PBX.

We have two main 
extensions and two ring groups. The first ring group rings the two internal 
extensions. If the internal extensions do not pick up the call after 15 seconds 
then the second ring group kicks in which should ring the two internal 
extensions plus two external numbers.

Firstly, how do I 
pass the DID number of an incoming call to the external number so that the 
external number sees the incoming number and not the voip dial out 
number?

Secondly, when the 
second ring group kicks in only one of the external numbers dials when both 
internal extensions and both external numbers should ring according to the ring 
group setting. Any ideals what's going wrong?

Kind 
regards

Brian. 
UK
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Re: [Asterisk-Users] Snom 360 Passsword Issue

[Tommaso Calosi]

I have had the same problem too,  I solved resetting the phone to 
factory defaults



Edward de Zeeuw wrote:

I'll take a look first thing tomorrow and let you know what I find.  Thanks!
Edward

Colin Anderson wrote:
  

In the Snom web management page under Advanced make sure Challenge response
on phone is turned to OFF. This is a stupid feature to have on by default
from the factory. 


-Original Message-
From: Edward de Zeeuw [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Snom 360 Passsword Issue


I have multiple (20+) Snom 360 phones communicating with asterisk
1.2.7.1.  Almost regularly (daily) and in some cases ongoing 9every 10
minutes the phones ask for password and id the account they are seeking
the password for.  If I hit the X key the phone continues operating
normally.  Has anyone else come across a similar issue?

Edward
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Re: [Asterisk-Users] 1.2.9.1 crashed today

[Steve Totaro]

Kevin P. Fleming wrote:

- Steve Totaro [EMAIL PROTECTED] wrote:
  

My system never stops processing calls but it slows.  On the CLI, it
may 
take five minutes for a command to execute but it does finally execute


(such as sip show peers or show channels)  And calls to the agents sit

in queue for a while before being delivered to an agent even thought
he 
agent is available.



There is a problem that has been identified that was caused by a recent change 
in Asterisk, that may be the reason you are seeing these symptoms. It would 
appear to only affect people using chan_agent, and then only if the agents are 
transferring calls they receive.

I am still working on a proper solution for this issue, but hope to get it 
resolved in the next day or so. Stay tuned and watch for a new release :-)

  

Kevin,

Thanks for the update.

We are a busy call center and asterisk just stopped running yesterday at 
a fairly peak hour.  We dropped 95 calls.  Asterisk just needed to be 
restarted.  This morning I am changing to boot to safe_asterisk.  Then 
we will have a core dump and asterisk should restart if it happens again. 

There is absolutely no reason in any of the system or asterisk logs 
explaining the crash. 

We do not do any transfers between agents but use queues and chan_agent 
extensively, we also use the manager interface to update agent penalties 
every fifteen minutes (as well as other functions).  We use alot of 
.call files to originate calls as well.  Hopefully this next release 
might help with our issue as well.


Thanks,
Steve
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Re: RE : Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[Steve Totaro]

You should have been working with asterisk about four years ago, lol!  
Talk about no docs.


Most opensource projects provide read between the lines and Google 
the error to get a clue, your brains, trial and error docs.  At least 
in my experience. 

Want some good docs?  Maybe read the docs for MySQL server,  FreeRadius 
server, Cisco gateways, MediaProxy or SIP Express Router and then retry 
CDRTool. 

Don't slam someone's project because you cannot figure it out and are 
not willing to pay $$$!  Comes off as greedy to me. 

Guess I have not setup the Harry Potter to junk folder rule on this 
machine yet, thanks for reminding me.


Thanks,
Steve


[EMAIL PROTECTED] wrote:

Steve,

I do hope ag-projects to keep their products with
commercial licences !

Harry

PS:

most of billing (or not) open source projects provide
a good documentation .
  



--- Steve Totaro [EMAIL PROTECTED] a
écrit :

  

Alex Robar wrote:


Harry,

Fisrt ag-projects talk about is product like a
  

gpl


software however they don't provide at least
  

some


documentation for non commercial users .


Are they required to release docs? It's a nice
  
bonus, but there's 


nothing forcing them to. If you want docs, you
  
could always be helpful 


and write some yourself.

try to call them !!
i'll offer you some money .

You can not Call them for some advices ... 



And why should you be able to? Again, it's nice to
  
have phone support, 


but in no way is it required. ViciDial doesn't
  
have phone support, but 


people use it. FreePBX doesn't offer phone
  
support, but it's pretty 


popular.

It's really a bad product don't waste your
  

time to


setup it.
this enterprise must be  fogotten it's
  

ag-projects .


it's not a reliable society ... more and more


Sounds to me like complaints from someone who
  
couldn't figure out how 


to get the product setup to me.

If you have valid complaints then by all means,
  
post it to the list as 


a warning... But even then, there's no need to
  
openly bash a product 


or company you don't like. This isn't the forum
  

for such things.


Alex

--
Alex Robar
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
  

 From their README

Disclaimer -- The information provided by
CDRTool documentation 
is not always enough to successfully complete the
installation and 
deployment of CDRTool. Most of the configuration
tasks are related to 
setting up components outside CDRTool environment.
Configuration of 
components like MySQL server, FreeRadius server,
Cisco gateways, 
MediaProxy or SIP Express Router require a project
management approach 
in place of a step by step installation procedure.


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Re: [Asterisk-Users] Hitting * in a queue call hangs up?

[Dinesh Nair]

On 06/20/06 18:20 Matt said the following:
It seems 1.2.9.1 does not correct this behavior... can I correct it 
somehow?


matt, i believe i've already sent this to the list.

the bug at http://bugs.digium.com/view.php?id=6897 has the fix for 1.2.x as 
agent-endcall.patch. apply that, and hitting '*' during a queue call wont 
hang up the call. to hangup the call you'd then need to use whatever was 
defined for disconnect in features.conf. also not that endcall=no in 
agents.conf.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
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RE: [Asterisk-Users] SE Michigan asterisk users group

[Watkins, Bradley]

I'd certainly be up for it, even if it ended up being a small group that
met over beer at a local pub. :)

Regards,
- Brad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz,
Steven
Sent: Thursday, June 22, 2006 4:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SE Michigan asterisk users group

I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.

How much interest in asterisk in Michigan is there on this list?

I am already on the board of glimasoutheast, with is a group for
technology professionals. (very broad range) It is a spin-off from
Automation Alley, which is SE Michigan's version of Silicone Valley.

--
Steven

http://www.glimasoutheast.org 

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RE: [Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while linkstate 0

[Watkins, Bradley]

I would say it's almost certainly a cabling issue of some sort.  We get
the S-frame while link down message all the time on our Asterisk
clusters due to the way the T1 failover switch works.  It's harmless in
our case since really the PRI is connected to the other box in the
cluster (unless there is a card failure, of course).  But for your
application, I'm sure you'd like it to be up. :)

Regards,
- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Stuart
Sent: Thursday, June 22, 2006 6:55 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE405P Dropping Calls. !! Got I-frame while
linkstate 0

Hello we have an * setup using a TE405P with a crossover to a Dialogic
PRI card. The machine with the Dialogic card is an automated call
generator. The asterisk machine is being used as a test platform to
answer the incoming calls play an audio file then record the incoming
audio. This set up has been working nicely when the calls were coming in
on a SIP channel.
However when call come in on the PRI it seems that * randomly drops all
channels. Some times every 30 seconds sometimes every 10 minutes. 
The * CLI will get the errors shown below. I have seen similar postings
about this error but found no explanation or solution.

The asterisk machine has no shared interrupts.

Could this be a cabling issue?

Write to 36 failed: Unknown error 500
Short write: 0/15 (Broken pipe)
 -- Executing Hangup(Zap/2-1, ) in new stack !! Got I-frame
while link state 0
 -- Hungup 'Zap/2-1'
!! Got S-frame while link down
!! Frame got rejected!
   == Primary D-Channel on span 1 up
 -- Executing Hangup(Zap/1-1, ) in new stack



Brian Stuart
Edulink Systems, Inc.
1(888)338-7177 ext. 219
http://www.edulinksys.com/ 


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[Asterisk-Users] best hardphone for Asterisk?

[Crazy Boy]

Dear Friends,We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost?Thank you.Regards,Chandra. 
		Ring'em or ping'em. Make  PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.___
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[Asterisk-Users] Call accounting where calls cross charge zones (code fragment request)

[Chris Bagnall]

Greetings list,

Before I go and write something from scratch, are there any kind souls here
who already have a nice code fragment that works our charging for calls
split across charging zones?

There are essentially 4 possibilities for a call:
1) call is completely within one zone, so it's nice and simple.
2) call starts in peak and finishes in the immediately following cheap zone.
3) call starts in cheap and finishes in the immediately following peak zone.
4) call is ridiculously long and completely encompasses one or more zones
(very unlikely, but probably should be able to handle it nonetheless).

I'm in the process of writing a billing module that works with asterisk's
own CDRs (or CDRs provided by an upstream provider) and doesn't use any
AGIs, etc.. A couple of quick questions:
1) does something like this already exist?
2) if not, would it be of any interest to fellow list readers?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] fax

[Khaled Chehab]

How can I support fax at trixbox 















M. Khaled
Chehab

Monitoring  Operationg Engineer 

Xplorium

Tel: +961 1
868686

Fax: +961 1
808810

e-mail: [EMAIL PROTECTED]








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This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

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[Asterisk-Users] asterisk sip listening port

[Khaled Chehab]

How I can let asterisk listen only at port 5062 since I have
ser on the same machine listening to port 5060 ,



Please from where I can configure it 












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Re: [Asterisk-Users] Routing inboud from ISDN to second * server.

[Thomas Laurids Pedersen]

I dont know if the extensions.conf is any use here as I am configuring the
access via freepbx ? pls let me know if so and I will post the file.

Other thing I am thinking of -  have seen other posts about diferrence IP
subnets must be configured in asterisk to allow thise calls to be routed.
Is this a requirement ?

Regards, Thomas



   
 Armin Schindler   
 [EMAIL PROTECTED] 
   To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 23-06-2006 07:30  Re: [Asterisk-Users] Routing inboud 
   from ISDN to second * server.   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




On Fri, 23 Jun 2006, Thomas Laurids Pedersen wrote:
 Hi All,

 I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2
 trunk between the 2 servers using the guide on dumbme. Trunk is not using
 register string and no authentication.

 In my dial plan I have 7XX numers on server B and 6xx numbers on Server
A.

 Calls from my SNOM phones are ok between the extensions on the 2 servers.

 In server A I have a eicon 4 port BRI card connected. Calls from outside
to
 extensions on server A works fine. (Inbound routes are configured for
 local extensions).

 Now how do I configure so that extensions on server B can be called from
 the ISDN connections ? like this ISDN - ServerA   -(IAX2)- ServerB -
 extension

 Server A does not have direct network access to the SNOM phone.

 From my reading I think that I have to make an inbound route or asterisk
 will not treat the call at all. This is also what I see if I debug capi.

 I have tried to make an inbound route and custom extension for all 7XX
 extentions and then entering the dial string IAX2/astB/7255 for extention
 725. Also tried the dial string IAX2/[EMAIL PROTECTED] with no luck.

 How do I do this ?

How does your extensions.conf look like?
I don't understand the problem. There is no difference between
isdn/capi calls or SNOM calls here, except if you have other ISDN numbers
you want to 'map' to the SNOM phones. The Dial() commands should look like
the same as for the other astA - astB connections.
Maybe you can give a real example.

Armin

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Re: [Asterisk-Users] Snom 360 with Firmware 6.1?

[Dr. Michael J. Chudobiak]

Koopmann, Jan-Peter wrote:

Hi,

Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I
am currently using Firmware 5.5 without serious problems but wanted to make
sure 6.X will work as well (including subscription etc.)


Use the very latest - 6.2.1. It seems quite good. Earlier versions 
(including 6.2.0) had problems.


- Mike
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Re: [Asterisk-Users] asterisk sip listening port

[Yair Hakak]

in the [general] section of sip.conf
bindport=5062

well documented here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf

-yair

On 6/23/06, Khaled Chehab [EMAIL PROTECTED] wrote:




How I can let asterisk listen only at port 5062 since I have ser on the same machine listening to port 5060 ,

Please from where I can configure it 




*No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.
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RE: [Asterisk-Users] best hardphone for Asterisk?

[Christian Stredicke]

snom 300 :-)

CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
  BoySent: Friday, June 23, 2006 7:16 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] best 
  hardphone for Asterisk?
  Dear Friends,We have implemented "Asterisk" in our 
  organization. There are 150 members in our organization. At present all are 
  using softphones. Now, I want to buy hardphones for our staff. Can anybody 
  suggest me that what is the best hardphone for Asterisk with 
  low-cost?Thank 
  you.Regards,Chandra.
  
  
  Ring'em or ping'em. Make PC-to-phone 
  calls as low as 1¢/min with Yahoo! Messenger with 
Voice.
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[Asterisk-Users] Antek EGW-804 e *

[Stefano Giuffredi]

Hi everybody,

I found in the company where I work an Antek EGW-804.

I googled to see if it can be configured to work with * and I understood that 
it is possible, but I don't know how.

Can someone help me?

Thanks
Stefano
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Re: [Asterisk-Users] fax

[JD Austin]

That's a vague question Khaled :)
First you must have hardware/software to support fax.
Faxes on my TDM400P work great with asterisk; they didn't work so great
over voip or with my X100P though.
Next you need software.. spandsp works ok to me with my fax to email
setup.
Read all about it here:
http://www.voip-info.org/wiki-Asterisk+fax

JD
Khaled Chehab wrote:

  
  
  
  
  
  How can I support fax at
trixbox 
  
  
  
  
  
  
  
  M. Khaled
Chehab
  Monitoring  Operationg
Engineer 
  Xplorium
  Tel: +961 1
868686
  Fax: +961 1
808810
  e-mail: [EMAIL PROTECTED]
  
  
  
  
  
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an
individual in this electronic message do not necessarily reflect views
of Xplorium or its subsidiaries and associates.
  
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from
disclosure belonging to Xplorium.
  
If you are not the intended addressee of this electronic message and
its attachments, kindly delete it immediately from your system and
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Xplorium does not guarantee the integrity of this electronic message
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or other defects.
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[Asterisk-Users] Realtime voicemail not working

[Mike]

Hi,

I've been using MySql with the CDR for awhile with no issues at all, but I
figured I'd try putting voicemail users in a DB.  Using the same DB (very
low load), and a user that has been proven to work well using a client GUI,
I inserted one user and tried to use realtime.  This is what I get in the
console:

ERROR[30517]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed
to connect database server asterisk on localhost. Check debug for more info.

I am using 1.2.4.  Where should I look for more detailed explanation than
failed to connect?

Mike

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[Asterisk-Users] Setting caller-id when parking call

[Matt]

I have an issue where someone will park a call, and then it will ring
back to them, but because the caller-id looks like a regular inbound
call, they don't know how to answer the call (these are the
receptionists).

I've tried to make an extention that I can transfer to that will set
the caller-id, and that works, but I'm having issues.

For instance if I do a blind transfer to an extention the CALLER
hears '71' instead of the receptionist.. DOH!  That didn't work.

If I do an attenteded transfer, then the caller-id gets set, but when
the person (that being the receptionist) hangs up, it gets unset.
DOH!  That didn't work either
now what? :)
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Re: [Asterisk-Users] best hardphone for Asterisk?

[Christian Victor]

Crazy Boy schrieb:
 We have implemented Asterisk in our organization. There are 150 members in 
 our organization. At present all are using softphones. Now, I want to buy 
 hardphones for our staff. Can anybody suggest me that what is the best 
 hardphone for Asterisk with low-cost?

I would say a Swissvoice IP 10S, a Snom 300 or - if you want better
quality - a Polycom 300.

The Snom looks good and is solid, the Swissvoice is similar plus it
supports PoE, the Polycom is a bit more expensive but worth the
additional cost.

Chris
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[Asterisk-Users] Trunk failover

[Mimmus]

Hi,
I'm doing some experiments with SIP vs. IAX2 trunks as an alternative to
PSTN and I noticed that, if voip link is down, failover to PSTN is almost
immediate with the SIP trunk and VEEERY slow with the IAX trunk.
Is there a specific reason? Some timeout to set in iax.conf?

Thanks
-- 
Domenico Viggiani

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[Asterisk-Users] calling between contexts

[René Enskat [Teamware GmbH]]

hi
all,

somebody know a way
how to call between contexts which are in a realtime
database?

i tried to include
them wise versa in extension.conf but this is not working.
Is there another
way?

regards
rene

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Re: [Asterisk-Users] best hardphone for Asterisk?

[Christian Victor]

Christian Stredicke schrieb:
 snom 300 :-)

Could be a bit hard to get 150 of them at one time imho. ;-)

Chris
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Re: [Asterisk-Users] freepbx centos 4 install script?

[Warren]

Michael Collins wrote:

  
Has anyone created a script that will download and install all of the
freepbx prerequisites in the INSTALL file automatically on a Centos 4

  
  box?
  
  In a manner of speaking the trixbox guys have.  Have you ever seen that
(or Asterisk @ Home)?  There is a script, install.sh, that installs a
bunch of stuff.  The FreePBX pre-reqs are mixed in with everything else,
but if you see the "yum -y install" line you'll see a ton of RPMs that
get installed, some of which are your pre-reqs.  Later on in the script
it has the actual installation - make, make install, etc.

Check it out: trixbox.org

-MC
  

I'm curious what "a manner of speaking" is. If I go that route what am
I losing? I really just want to make sure whichever route I go I will
be able to come here for help and not get blown off because of
something non-standard in the packaging I chose.

W


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[Asterisk-Users] Dial(ZAP with t option for call transfer via *2)

[Bruno . Voigt]

Hi all,

using asterisk 1.2.9.x I would like to forward an incoming call to
an outbound ZAP target (EuroISDN PRI via Digium TE410P),
i.e. an mobile phone.

exten = 105,1,Dial(Zap/r1/0171234567,120,rt)

I use the Dial() option t
as the goal is to enable the called destination to be able to perform
a attended call transfer via *2 to another extension of the PBX.

But asterisk doesn't seem to screen the outbound leg of type Zap channel
for *2 - it is just ignored.

Using a SIP - Target in the above Dial() statement
the reached target is able to use *2.

What is the current status of this feature in asterisk?
What is needed to make it work with Zap-Channels ??

features.conf
[featuremap]
atxfer = *2; Attended transfer

TIA,
Bruno
--
[EMAIL PROTECTED]
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[Asterisk-Users] UK English Sounds

[Steve Kennedy]

New versions of the Male UK English sound files are now available.

We believe these are complete for v1.2.x of Asterisk and v1.2.1 of
Asterisk-sounds.

The LouisLouis song is missing (which is US anyway) and 7 seconds of
silence, but everything else should be there. The vm voice prompts are
now correct too.

There are two additional sets of files available for download

LineTones contains tones (of 20Hz to 20KHz and one that steps through
them) - this is available in gsm, pcm and wav format.

UKCounties contains the county names for UK (England, Scotland and
Wales).

They are all available from http://www.tel.net/


Thanks again to Jay [EMAIL PROTECTED] for recording them and Jim
[EMAIL PROTECTED] for doing the post work.



Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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RE: [Asterisk-Users] best hardphone for Asterisk?

[jacobso1]

If you do not have a budget, grandstream is not bad

You do not get what you do not pay
But you do not allways get what you paid for

t. Jacobson 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Victor
Sent: vendredi 23 juin 2006 14:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] best hardphone for Asterisk?

Crazy Boy schrieb:
 We have implemented Asterisk in our organization. There are 150 members
in our organization. At present all are using softphones. Now, I want to buy
hardphones for our staff. Can anybody suggest me that what is the best
hardphone for Asterisk with low-cost?



-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.9.2/373 - Release Date: 22/06/2006
 

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Re: [Asterisk-Users] Out of Office Auto Reply:

[Mitch Thompson]

Dean Collins wrote:


You got to be freaking kidding, a month of this?
Cant we get an easy process for the list owner to take care of these?





 


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


[EMAIL PROTECTED]
 


Sent: Thursday, 22 June 2006 11:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Out of Office Auto Reply:


I will be on vacation from 22/06/06 to 30/06/06.

I will not be reachable on my mobile. I will have limited access to
   


m


What, a month is only 8 days long where you live?

--
To sit home, read one's favorite paper, and scoff at the misdeeds of 
the men who do things is easy, but it is markedly ineffective. It is 
what evil men count upon the good men's doing. —Theodore Roosevelt


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Re: [Asterisk-Users] best hardphone for Asterisk?

[Andrew Latham]

I have not had a problem getting SNOMs.  Keep in mind that SNOM phones
have a penguin inside.  m,,  Penguins..

On 6/23/06, Christian Victor [EMAIL PROTECTED] wrote:

Christian Stredicke schrieb:
 snom 300 :-)

Could be a bit hard to get 150 of them at one time imho. ;-)

Chris
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [Asterisk-Users] Dell PowerEdge 1650

[Warren]

Sean,

I do not have one up and running, however I do remember seeing various
posts earlier this year about incompatibilities with Dell motherboards
and Digium hardware.  IIRC the solution was to either change to a
different server or to change to a Sangoma board.

You might want to search the archives on this.

W

Sean Cook wrote:

 Anyone have a 1650 running successfully in production mode with 2-4
 PRI's?  I want to make sure I don't have a motherboard compatibility
 problem before I buy one of these.  We are going to be using a Digium
 TE210P to start off with and probably moving to the TE411P down the
 road aways?

 thanks,

 Sean


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[Asterisk-Users] SIP - PSTN calls not connecting properly

[Ronan Mullally]

Hi,

I've got a problem with my asterisk set up which has been going on for a
while (months).  I'm currently running 1.2.7.1 on a gentoo box with the
topology below:


 +-+
   PSTN -+  *  +- Service Provider
   (wctdm400p)   +-+-+-+ IAX
   | |
   | |
 FXS --+ +-- SIP (cisco 7940)


I can make calls from the FXS port to the PSTN or my IAX service provider
without any problems.

I can make calls from my SIP phone to my IAX service provider, also without
any problems.

I can receive calls to the FXS port and SIP phone without any problems.

However, when I call from my SIP phone to the PSTN my calls die, repeatedly,
after 2-3 minutes.  The display on the phone shows 'Session Progress (in
183)' for the duration of the call, rather than 'Connected', so it looks
like the SIP phone is not recognising call connection on the PSTN.

Output from the console is as follows:

-- Executing Dial(SIP/ronan-5e0e, Zap/4/xxx) in new stack
-- Called 4/xxx
-- Hungup 'Zap/4-1'
  == Spawn extension (default, xxx, 1) exited non-zero on 
'SIP/ronan-5e0e'

A packet trace from the * box shows:

 ...

 16.758516  192.168.2.9 - 192.168.2.30 UDP Source port: 12230  Destination 
port: 31042
 16.758595 192.168.2.30 - 192.168.2.9  UDP Source port: 31042  Destination 
port: 12230
 16.778540  192.168.2.9 - 192.168.2.30 UDP Source port: 12230  Destination 
port: 31042
 16.779004 192.168.2.30 - 192.168.2.9  UDP Source port: 31042  Destination 
port: 12230
 16.790884 192.168.2.30 - 192.168.2.9  SIP Request: CANCEL sip:[EMAIL 
PROTECTED];user=phone
 16.791266  192.168.2.9 - 192.168.2.30 SIP Status: 487 Request Terminated
 16.791477  192.168.2.9 - 192.168.2.30 SIP Status: 200 OK

(192.168.2.9 is the * box, .30 is the phone)

This has been going on for some time, but I've put up with it as the
majority of my calls are short so it's not a big issue.  As a result I'm
unsure when the problem started, so I've no idea what change I made to the
config that caused it.  I'm fairly sure the change is on asterisk as I've
not touched the config on the 7940 in a long time.

My zaptel.conf, zapata.conf and sip.conf files are below, any suggestions or
clue transfer would be much appreciated.


-Ronan

# zaptel.conf
loadzone=uk
defaultzone=uk
fxsks=4
fxoks=1-3

# zapata.conf
[channels]
group = 0
context = incoming-POTS
signalling = fxs_ks
rxgain=10.0
txgain=6.0
echocancel=yes
echocancelwhenbridged=no
echotraining=300
immediate=no
busydetect=no
busycount=5
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
callprogress=yes
callwaiting=yes
relaxdtmf=no
progzone=uk
useincomingcalleridonzaptransfer = yes
usecallerid=no
callerid=asreceived
cidsignalling=v23
cidstart=polarity
ukcallerid=yes
channel = 4

# sip.conf
[general]
allow=ulaw
allow=alaw
allow=gsm
allow=g723.1
context=incoming
recordhistory=yes
port=5060
bindaddr=0.0.0.0
srvlookup=yes
tos=lowdelay
defaultexpirey=120
nat=no
localnet=192.168.0.0/255.255.252.0

[ronan]
regextension=ronan
regcontext=4L
[EMAIL PROTECTED]
callerid=Ronan Mullally 100
restrictcid=no
callgroup=1,2
pickupgroup=1,2
host=dynamic
language=en
type=friend
context=default
username=ronan
secret=x
fromdomain=4L.ie
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g723.1
qualify=100
accountcode=ronan
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Re: [Asterisk-Users] Realtime voicemail not working

[Benjamin Stocker]

2006/6/23, Mike [EMAIL PROTECTED]:
Hi,I've been using MySql with the CDR for awhile with no issues at all, but Ifigured I'd try putting voicemail users in a DB.Using the same DB (verylow load), and a user that has been proven to work well using a client GUI,
I inserted one user and tried to use realtime.This is what I get in theconsole:ERROR[30517]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failedto connect database server asterisk on localhost. Check debug for more info.
What's in your res_mysql.conf file? You need to place the connection settings there like you did in cdr_mysql.conf. If you still get errors, start asterisk using the -d option and check /var/log/asterisk. Also check 
logger.conf to make sure debugging is enabled.
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Re: [Asterisk-Users] calling between contexts

[Benjamin Stocker]

2006/6/23, René Enskat [Teamware GmbH] [EMAIL PROTECTED]:







hi 
all,

somebody know a way 
how to call between contexts which are in a realtime 
database?

i tried to include 
them wise versa in extension.conf but this is not working.
Is there another 
way?I solved this using the Goto and GotoIf commands.
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Re: [Asterisk-Users] best hardphone for Asterisk?

[Joshua West]

I find the Polycom Soundpoint 301 and 501 models to be great phones.

Christian Victor wrote:
 Crazy Boy schrieb:
   
 We have implemented Asterisk in our organization. There are 150 members in 
 our organization. At present all are using softphones. Now, I want to buy 
 hardphones for our staff. Can anybody suggest me that what is the best 
 hardphone for Asterisk with low-cost?
 

 I would say a Swissvoice IP 10S, a Snom 300 or - if you want better
 quality - a Polycom 300.

 The Snom looks good and is solid, the Swissvoice is similar plus it
 supports PoE, the Polycom is a bit more expensive but worth the
 additional cost.

 Chris
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-- 
Joshua West
Linux Infrastructure Engineer
Boston Engineering Corporation
http://www.boston-engineering.com


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Re: [Asterisk-Users] problem - DSL line and Digium card

[Eric Hartley]

 I was wondering if anyone
has seen a similar problem...

  
 I have a DSL line that doubles as a voice line to my Asterisk box.  
 When the Digium card answers that line, the DSL modem is disconnected 
 until the line releases.  The phone line is split at the DSL modem 
 then run to an FXO port.

 I'm using one of the 4 port analog Digium cards (3 FXO/1 FXS).
  

Do you have a line filter on the FXO port?


Currently no.  That is my first plan of action.  My concern is that the Digium card might be pulling too much current and not leaving enough for the  DSL modem.

Eric




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Re: [Asterisk-Users] Dell PowerEdge 1650

[Matt]

I can't comment on the 1650, but we are running three 2850s in
production, with heavy load, and they are working fine.

In one I have 3 PRI cards, in another, only 1.

On 6/22/06, Sean Cook [EMAIL PROTECTED] wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Anyone have a 1650 running successfully in production mode with 2-4
PRI's?  I want to make sure I don't have a motherboard compatibility
problem before I buy one of these.  We are going to be using a Digium
TE210P to start off with and probably moving to the TE411P down the
road aways?

thanks,

Sean
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.3 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFEmwOb1Kolm8VQlAURAtBOAJsFdirbMhilGmcKd07+tORXxwZLKgCfQrjs
3hoZGLFllzW6xLrpRuuxjMk=
=3bOb
-END PGP SIGNATURE-

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Re: [Asterisk-Users] asterisk sip listening port

[Michiel van Baak]

On 14:42, Fri 23 Jun 06, Khaled Chehab wrote:
 How I can let asterisk listen only at port 5062 since I have ser on the same
 machine listening to port 5060 ,
 
  
 
 Please from where I can configure it 

sip.conf

-- 
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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[Asterisk-Users] Kernel 2.4 / 2.6 and timer

[Daniel Salama]

I've read in different places that if I want to do trunking and  
meetme on Asterisk I need to have a reliable timer. People have  
recommended that I install a Digium board, even if I don't have any  
circuits connected to it, just to get a reliable timer. However, I've  
also read that if I'm using kernel 2.6, I don't need to have a Digium  
board.


I have a few servers that need to do trunking and meetme and I don't  
have ANY PSTN-type circuits. I do everything via VoIP. All my servers  
are running kernel 2.6. Do I really need to have these boards?


Thanks,
Daniel
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Re: [Asterisk-Users] SIP - PSTN calls not connecting properly

[Brian Swan]

I had this same problem.  For me, the Cisco phone wasn't detecting  
that the call was connected.  Turn on VAD, and maybe bump up the rx  
gain on the PSTN.


Hope that helps,
Brian

On Jun 23, 2006, at 8:04 AM, Ronan Mullally wrote:


Hi,

I've got a problem with my asterisk set up which has been going on  
for a
while (months).  I'm currently running 1.2.7.1 on a gentoo box with  
the

topology below:


 +-+
   PSTN -+  *  +- Service Provider
   (wctdm400p)   +-+-+-+ IAX
   | |
   | |
 FXS --+ +-- SIP (cisco 7940)


I can make calls from the FXS port to the PSTN or my IAX service  
provider

without any problems.

I can make calls from my SIP phone to my IAX service provider, also  
without

any problems.

I can receive calls to the FXS port and SIP phone without any  
problems.


However, when I call from my SIP phone to the PSTN my calls die,  
repeatedly,
after 2-3 minutes.  The display on the phone shows 'Session  
Progress (in
183)' for the duration of the call, rather than 'Connected', so it  
looks

like the SIP phone is not recognising call connection on the PSTN.

Output from the console is as follows:

-- Executing Dial(SIP/ronan-5e0e, Zap/4/xxx) in new  
stack

-- Called 4/xxx
-- Hungup 'Zap/4-1'
  == Spawn extension (default, xxx, 1) exited non-zero on  
'SIP/ronan-5e0e'


A packet trace from the * box shows:

 ...

 16.758516  192.168.2.9 - 192.168.2.30 UDP Source port: 12230   
Destination port: 31042
 16.758595 192.168.2.30 - 192.168.2.9  UDP Source port: 31042   
Destination port: 12230
 16.778540  192.168.2.9 - 192.168.2.30 UDP Source port: 12230   
Destination port: 31042
 16.779004 192.168.2.30 - 192.168.2.9  UDP Source port: 31042   
Destination port: 12230
 16.790884 192.168.2.30 - 192.168.2.9  SIP Request: CANCEL  
sip:[EMAIL PROTECTED];user=phone
 16.791266  192.168.2.9 - 192.168.2.30 SIP Status: 487 Request  
Terminated

 16.791477  192.168.2.9 - 192.168.2.30 SIP Status: 200 OK

(192.168.2.9 is the * box, .30 is the phone)

This has been going on for some time, but I've put up with it as the
majority of my calls are short so it's not a big issue.  As a  
result I'm
unsure when the problem started, so I've no idea what change I made  
to the
config that caused it.  I'm fairly sure the change is on asterisk  
as I've

not touched the config on the 7940 in a long time.

My zaptel.conf, zapata.conf and sip.conf files are below, any  
suggestions or

clue transfer would be much appreciated.


-Ronan

# zaptel.conf
loadzone=uk
defaultzone=uk
fxsks=4
fxoks=1-3

# zapata.conf
[channels]
group = 0
context = incoming-POTS
signalling = fxs_ks
rxgain=10.0
txgain=6.0
echocancel=yes
echocancelwhenbridged=no
echotraining=300
immediate=no
busydetect=no
busycount=5
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
callprogress=yes
callwaiting=yes
relaxdtmf=no
progzone=uk
useincomingcalleridonzaptransfer = yes
usecallerid=no
callerid=asreceived
cidsignalling=v23
cidstart=polarity
ukcallerid=yes
channel = 4

# sip.conf
[general]
allow=ulaw
allow=alaw
allow=gsm
allow=g723.1
context=incoming
recordhistory=yes
port=5060
bindaddr=0.0.0.0
srvlookup=yes
tos=lowdelay
defaultexpirey=120
nat=no
localnet=192.168.0.0/255.255.252.0

[ronan]
regextension=ronan
regcontext=4L
[EMAIL PROTECTED]
callerid=Ronan Mullally 100
restrictcid=no
callgroup=1,2
pickupgroup=1,2
host=dynamic
language=en
type=friend
context=default
username=ronan
secret=x
fromdomain=4L.ie
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g723.1
qualify=100
accountcode=ronan
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RE: [Asterisk-Users] Realtime voicemail not working

[Mike]

That is what I have in my config file. Pretty 
much the same thing as the CDR file

[general]dbhost=localhostdbname=dbdbuser=userdbpass=pw;dbport=3306;dbsock = 
/tmp/mysql.sock


Mike



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin 
StockerSent: June 23, 2006 9:07 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Realtime voicemail not working

2006/6/23, Mike [EMAIL PROTECTED]:
Hi,I've 
  been using MySql with the CDR for awhile with no issues at all, but 
  Ifigured I'd try putting voicemail users in a DB.Using the 
  same DB (verylow load), and a user that has been proven to work well using 
  a client GUI, I inserted one user and tried to use 
  realtime.This is what I get in 
  theconsole:ERROR[30517]: res_config_mysql.c:615 mysql_reconnect: 
  MySQL RealTime: Failedto connect database server asterisk on localhost. 
  Check debug for more info. What's in your 
res_mysql.conf file? You need to place the connection settings there like you 
did in cdr_mysql.conf. If you still get errors, start asterisk using the -d 
option and check /var/log/asterisk. Also check logger.conf to make sure 
debugging is enabled.
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[Asterisk-Users] Meetme max users

[Bartosz Wegrzyn - asterisk]

Does anyone knows what is the max of users that meefme can handle.
I am using Iax2 clients to connect to the conference.

Thanks


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[Asterisk-Users] Asterisk 1.4 on schedule?

[Obelix]

Is Asterisk 1.4 on schedule for release in July?

/Obelix
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RE: [Asterisk-Users] Realtime voicemail not working

[Mike]

Weird. I just tried with 127.0.0.1 instead of 
localhost and it worked. Can't explain why

How can I force it to go through a db socket 
instead?

Mike


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin 
StockerSent: June 23, 2006 9:07 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Realtime voicemail not working

2006/6/23, Mike [EMAIL PROTECTED]:
Hi,I've 
  been using MySql with the CDR for awhile with no issues at all, but 
  Ifigured I'd try putting voicemail users in a DB.Using the 
  same DB (verylow load), and a user that has been proven to work well using 
  a client GUI, I inserted one user and tried to use 
  realtime.This is what I get in 
  theconsole:ERROR[30517]: res_config_mysql.c:615 mysql_reconnect: 
  MySQL RealTime: Failedto connect database server asterisk on localhost. 
  Check debug for more info. What's in your 
res_mysql.conf file? You need to place the connection settings there like you 
did in cdr_mysql.conf. If you still get errors, start asterisk using the -d 
option and check /var/log/asterisk. Also check logger.conf to make sure 
debugging is enabled.
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RE: [Asterisk-Users] SE Michigan asterisk users group

[Tim Sharp]

Steven,
I am in Livonia.  Please let me know if there is interest.
Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BerkHolz,
Steven
Sent: Thursday, June 22, 2006 4:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SE Michigan asterisk users group


I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.

How much interest in asterisk in Michigan is there on this list?

I am already on the board of glimasoutheast, with is a group for
technology professionals. (very broad range)
It is a spin-off from Automation Alley, which is SE Michigan's version
of Silicone Valley.

-- 
Steven

http://www.glimasoutheast.org 

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Re: [Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings

[Erick Perez]

Tom, just to make sure im on the right track.
What files do you tweak?
sip.conf, the ones from avaya and anything else?


On 6/22/06, Tom Lynn [EMAIL PROTECTED] wrote:

Nope.  Let me know if you do.  I've suspended my efforts until I see a new
version of firmware available on the Avaya web site.


On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote:
 Thanks for your comments Tom. Indeed the MWI and the programmable
 buttons are the only things that do not work for me. Besides that, the
 phone is great and the audio quality is superb.
 Did you managed somehow to make the MWI work?

 Will keep searching the net, the 4602 page is somehow poor on the
documentation.

 On 6/21/06, Tom Lynn  [EMAIL PROTECTED] wrote:
  Well, I wouldn't say nobody.  I do and I've corresponded with a few
people
  that do. There's a page on voip-info.org dedicated to the Avaya 4602
  telephone and SIP (I'm hoping I'm not the only reader of that page).
When
  I've used my Avaya phone in conference (FWD CoffeeHouse), I've had
people
  sincerely compliment me on the quality of sound with my phone.
 
  But..
 
  Avaya has a few things working against it within the context of
Asterisk:
 
  * MWI just doesn't work (If you insist on trying it, get ready for your
  phone to lose it's registration with * every hour or so)
  * Dial strings beginning with * character appear to go nowhere with
these
  phones
  * They're perceived as rather expen$ive
  * As a company, they're simply not focused on * since it doesn't help
sell
  any of their other product.  They prefer selling things that drive
  maintenance contract revenue and, let's face it, the phone is the
commodity
  appliance that connects to *.  Even within the enterprise space, very
few
  carry maintenance on their telephone sets anymore.
 
  Funny anectdote:  Avaya loves showing Cisco 79xx phones with a SIP load
  registered to their PBX systems with a Powered By Avaya background.
They
  claim that, unlike Cisco, they will accept third party SIP clients
  registering to their system.  However, they really don't provide any
kind of
  support for their phones used with a system other than their own.  My
Mom
  used to call that the Pot calling the Kettle Black.
 
  Good phone, great sound, just no support and a bit wonky on the
features.
 
  My 2 cents.
 
 
 
 
 
 
  On 6/21/06, Erick Perez  [EMAIL PROTECTED] wrote:
  
  nobody uses avaya phones with asterisk?
 
  On 6/20/06, Erick Perez  [EMAIL PROTECTED] wrote:
   Hi, I setup my tftp to send SIP configurations (the bin files) to the
   avaya phone. When it finished loading and rebooting it asked for the
   extension and the password and the asterisk ip address. I had to input
   that manually and is now working perfectly with asterisk.
  
   what is the format of the text files to make this phone load the
   asterisk ip, extension number, codec used, password as well as to
   configure message waiting indicator and maybe modify some of the
   buttons (such as just pressing one of the available programmable
buttons to access voicemail). I have 10 more of these phones and i
   want to do provisioning automatically.
  
   in the 46xxsettings.txt file there are no such parameters
  
  
   thanks,
  
  
   --
  
 

   Erick Perez
   Panama Sistemas
   Integradores de Telefonia IP y Soluciones Para Centros de Datos
   Panama, Republica de Panama
   Cel Panama. +(507) 6694-4780
  
 

  
 
 
  --
 

  Erick Perez
  Panama Sistemas
  Integradores de Telefonia IP y Soluciones Para Centros de Datos
  Panama, Republica de Panama
  Cel Panama. +(507) 6694-4780
 

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 --


 Erick Perez
 Panama Sistemas
 Integradores de Telefonia IP y Soluciones Para Centros de Datos
 Panama, Republica de Panama
 Cel Panama. +(507) 6694-4780


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[Asterisk-Users] Re: Showing Current Calls

[Steven]

There is also:
show channels verbose
show channels concise

These may be easier for you to interpret.

-- 
-- 
Steven

http://www.glimasoutheast.org



Douglas Garstang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Can someone recommend the best way to view current calls in progress on the 
Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to read.

hestia*CLI show channels
Channel  Location State   Application(Data)
SIP/2944093-f9e2 (None)   Up  Bridged Call(SIP/2944079-e7f2)
SIP/2944079-e7f2 [EMAIL PROTECTED]:2  Up  Dial(SIP/2944093|36|tr)
2 active channels
1 active call

hestia*CLI
hestia*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last 
Message
xxx.yyy.128.115  (None)  e77bba33-cc  00101/02261  unkn  No   Rx: 
REGISTER
xxx.yyy.128.110  (None)  739f4603-e8  00101/00778  unkn  No   Rx: 
REGISTER
xxx.yyy.128.86   (None)  56caad3a-eb  00101/01046  unkn  No   Rx: 
REGISTER
xxx.yyy.128.115  (None)  91ea0410-60  00101/02262  unkn  No   Rx: 
REGISTER
xxx.yyy.128.86   (None)  488801e-105  00101/01046  unkn  No   Rx: 
REGISTER
xxx.yyy.128.86   (None)  c3b27274-ef  00101/01194  unkn  No   Rx: 
REGISTER
xxx.yyy.128.77   2944093 2405f1ef74d  00102/0  ulaw  No   Tx: ACK
xxx.yyy.128.83   2944079 cf1722ef-cc  00101/2  ulaw  No   Rx: ACK

Doug.




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[Asterisk-Users] Re: Out of Office Auto Reply:

[Steven]

Exchange changes

http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp

-- 
-- 
Steven

http://www.glimasoutheast.org



Koopmann, Jan-Peter [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote:

 We use MS Exchange too and, as far as I am aware, it is cognizant of
 mailing list headers and doesn't send OOO notices to mailing list
 postings. The only mailing list from which I receive my own OOO
 notices is one that doesn't have the proper mailing list headers set.


No. Exchange does not honour Precedence headers. It has some funky way of 
determining what is a mailing list and what is not. It 
does not work very well and it has (or had) to be enabled via a registry key. 
If you don't do this, even Exchange 2003 will reply to 
some mailing lists. But it should not send this to every mail but only once 
day...


Kind regards,
  JP

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Re: [Asterisk-Users] Kernel 2.4 / 2.6 and timer

[Filip Drągowski]

You don't need any boards. only zaptel and ztdummy module loaded and working
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
.../zaptel-1-2-X/README.udev
litte modyfication in ztdummy.c
that was all i needed
I've read in different places that if I want to do trunking and meetme 
on Asterisk I need to have a reliable timer. People have recommended 
that I install a Digium board, even if I don't have any circuits 
connected to it, just to get a reliable timer. However, I've also read 
that if I'm using kernel 2.6, I don't need to have a Digium board.


I have a few servers that need to do trunking and meetme and I don't 
have ANY PSTN-type circuits. I do everything via VoIP. All my servers 
are running kernel 2.6. Do I really need to have these boards?


Thanks,
Daniel


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Re: [Asterisk-Users] Realtime voicemail not working

[Benjamin Stocker]

2006/6/23, Mike [EMAIL PROTECTED]:





That is what I have in my config file. Pretty 
much the same thing as the CDR file

[general]dbhost=localhostdbname=dbdbuser=userdbpass=pw;dbport=3306
;dbsock = 
/tmp/mysql.sock

In logger.conf, enable debug mode: debug = notice,warning,error,debug,verboseStart asterisk in debug mode: asterisk -d
Check /var/log/asterisk/debug, you should see more details about the connection problem there.
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Re: [Asterisk-Users] Meetme max users

[Matt Florell]

We've had over 100 participants spread across 30 meetme rooms on a
single server before,  and the most we've had in a single meetme room
is 46. I don't know of a hard limit for meetme participants and I
haven't seen a limit in the code. You would most likely be limited by
the resources on your server I would guess.

MATT---

On 6/23/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote:

Does anyone knows what is the max of users that meefme can handle.
I am using Iax2 clients to connect to the conference.

Thanks


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[Asterisk-Users] How to use G729 decoded voice files?

[Obelix]

If you need to use prerecord voicer files for G729 codec, how do you configure
them?

Do they have to be specially named, copied to their own folder or something?

Can asterisk automatically find them even if you use standard names?

/Obelix
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RE: [Asterisk-Users] Realtime voicemail not working

[Adam Linford]

I think you just comment out dbhost
statement and leave the dbsock statement in, and it assumes its a localhost
connection on the socket specified.



Cheers,

Adam











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Mike
Sent: 23 June 2006 14:51
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Realtime voicemail not working





Weird. I just tried with 127.0.0.1
instead of localhost and it worked. Can't explain why



How can I force it to go through a db
socket instead?



Mike









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Stocker
Sent: June 23, 2006 9:07 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Realtime voicemail not working





2006/6/23, Mike [EMAIL PROTECTED]:


Hi,

I've been using MySql with the CDR for awhile with no issues at all, but I
figured I'd try putting voicemail users in a DB.Using the same DB
(very
low load), and a user that has been proven to work well using a client GUI, 
I inserted one user and tried to use realtime.This is what I get in
the
console:

ERROR[30517]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed
to connect database server asterisk on localhost. Check debug for more info. 




What's in your res_mysql.conf file? You need to place the connection settings
there like you did in cdr_mysql.conf. If you still get errors, start asterisk
using the -d option and check /var/log/asterisk. Also check logger.conf to make
sure debugging is enabled.








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[Asterisk-Users] Tribox - Unistim9.4 Makefile

[Ian Cowley]

Anyone had any luck getting the Unistim channel driver to install on
Tribox 1.0.5?

 
Ian Cowley
Network  Security

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Re: [Asterisk-Users] voip to voip bridge

[Benoît Mérouze]

Extracted from http://www.voip-info.org/wiki-Asterisk+cmd+Dial:

' When options /t/, /T, h, H, w, W or L (with multiple 
arguments) are applied, Asterisk will remain in the media path, even if 
/canreinvite=yes'' (a SIP channel option) has been specified.'


Then how is it possible to limit a call without the L option ?



Benoît Mérouze wrote:

Hi,

I've got some problems with bridged calls, the quality is extremely 
poor (more or less blanks or one way voice issues). But if I do a 
normal call with the same provider, there is no problem.


Reinvite is enabled, but what are the parameters in the dial command 
that force asterisk to stay in the loop ?
Are the H (to allow caller to hang up by dialing *) or L (to limit the 
call) parameters ones of them ?


As an example, here is a Dial command I execute to bridge a call to a 
new one :

SIP/kddi/0033172699611|30|HL(162:6:3)

Thanks,
Benoit



[EMAIL PROTECTED] wrote:


Hi,

Check if reinvites are enabled, and that you don’t use any parameter 
in the dial command that forces asterisk to stay in the loop.


Ohad



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Erick 
Baum

*Sent:* Wednesday, June 14, 2006 5:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] voip to voip bridge

Has anyone had any good experiences with a voip to voip bridge... 
where you have an incoming call on a voip line which is redirected 
out another voip line to a regular phone line? Whenever we do this, 
the connected call is kinda lagged and the quality isn't always that 
great. It seems to me this is just a problem with the inherent delay 
in the voip connections. But I was wondering if there's any special 
configurations that could make the situation better?


Erick






--
Benoît Mérouze
_._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._.
Groupe IPercom - The VoIP Enabling Company -  http://www.ipercom.com
Ingénieur RD - courriel : [EMAIL PROTECTED]
Network Software Developer - mailto: [EMAIL PROTECTED]
Tél. / Phone : +33 1 7269 9611
._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._
Siège Social 43, rue Fessart 92100 Boulogne Billancourt
RCS NANTERRE B 440 345 528 - Capital social: 100 000 €
CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES
SONT CONFIDENTIELS  COUVERTS PAR LE SECRET PROFESSIONNEL

THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE
CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY
._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._
Only two things are infinite, the universe and human stupidity, and I'm
 not sure about the former.
Albert Einstein

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RE: [Asterisk-Users] best hardphone for Asterisk?

[Jonathan k. Creasy]

I'll second that. I really like the provisioning features. My customers
prefer the 501 because they like the layout and speaker phone
functionality. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
West
Sent: Friday, June 23, 2006 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] best hardphone for Asterisk?

I find the Polycom Soundpoint 301 and 501 models to be great phones.

Christian Victor wrote:
 Crazy Boy schrieb:
   
 We have implemented Asterisk in our organization. There are 150
members in our organization. At present all are using softphones. Now, I
want to buy hardphones for our staff. Can anybody suggest me that what
is the best hardphone for Asterisk with low-cost?
 

 I would say a Swissvoice IP 10S, a Snom 300 or - if you want better
 quality - a Polycom 300.

 The Snom looks good and is solid, the Swissvoice is similar plus it
 supports PoE, the Polycom is a bit more expensive but worth the
 additional cost.

 Chris
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-- 
Joshua West
Linux Infrastructure Engineer
Boston Engineering Corporation
http://www.boston-engineering.com


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RE: [Asterisk-Users] Re: Out of Office Auto Reply:

[Colin Anderson]

Should be part of the FAQ for the list, as well as the setting for Exchange
5.5 which a *lot* of orgs still run (we do too)

I wonder if the list SW can be modded to automatically plonk any mail with
the subject string: Out of Office 


-Original Message-
From: Steven [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 8:08 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Out of Office Auto Reply:


Exchange changes

http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp

-- 
-- 
Steven

http://www.glimasoutheast.org



Koopmann, Jan-Peter [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote:

 We use MS Exchange too and, as far as I am aware, it is cognizant of
 mailing list headers and doesn't send OOO notices to mailing list
 postings. The only mailing list from which I receive my own OOO
 notices is one that doesn't have the proper mailing list headers set.


No. Exchange does not honour Precedence headers. It has some funky way of
determining what is a mailing list and what is not. It 
does not work very well and it has (or had) to be enabled via a registry
key. If you don't do this, even Exchange 2003 will reply to 
some mailing lists. But it should not send this to every mail but only once
day...


Kind regards,
  JP

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[Asterisk-Users] call quality statistics?

[Dr. Michael J. Chudobiak]

Is it possible to set up some sort of call-quality statistics 
reporting/logging for IAX2 calls? Something that can keep track of 
dropped packet / jitter trends?


(I know iax2 show channels shows this info for active calls.)

Suggestions appreciated!


- Mike

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RE: [Asterisk-Users] freepbx centos 4 install script?

[Michael Collins]

 

I'm curious what a manner of speaking
is. If I go that route what am I losing? I really just want to make
sure whichever route I go I will be able to come here for help and not get
blown off because of something non-standard in the packaging I chose.






By manner of speaking I mean
that the Trixbox install CD includes a script that installs FreePBX, among
other things. It doesnt have a script or instructions that say, Here
is an automated means to install FreePBX, here are the pre-requisites, etc.
So, in a manner of speaking there is information on installing FreePBX
from a script, its just that it is mixed in with a number of other
Asterisk-related goodies.



-MC








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[Asterisk-Users] Echocancelwhenbridged

[Wildheart]

Hi,

   Can someone tell me what are the valid parameters for the option
echocancelwhenbridged? Is it just yes or no, or does it support 128 as
well? Also is thier any differnce with using

   echocancelwhenbridged=128 as opposed to echocancelwhenbridged=yes
(assuming that 128 is a valid option).

   With thanks,

 Tim

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[Asterisk-Users] Asterisk Users Group - Portugal

[Marco Mouta]

Boa tarde,

Após alguma experiência com o Asterisk, e com muito ainda para
aprender, gostaria de saber se há alguém nesta mailing list que
pretenda criar um Asterisk Users Group para Portugal.

Visto que acaba sempre por ser uma enorme aprendizagem ( valor
acrescentado) a partilha de experiências/problemas e soluções nas
implementações Asterisk.

Há spre detalhes que variam entre os Telco's de cada país, voice prompts, etc.


Se houver um número minimo de pessoas interessadas, podemos avançar com a ideia.

--
Com os melhores cumprimentos,

Marco Mouta
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[Asterisk-Users] Passing DID to external number?

[Philippe Lindheimer]

You already posted this. I answered it yesterday also?p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 10:27:19 +0100Subject: [Asterisk-Users] Passing DID to external number?  Hi,  We run a small  switchboard using Asterisk and Free PBX.  We have two main  extensions and two ring groups. The first ring group rings the two internal  extensions. If the internal extensions do not pick up the call after 15 seconds  then the second ring group kicks in which should ring the two internal  extensions plus two external numbers.  Firstly, how do I  pass the DID number of an incoming call to the external number so that the  external number sees the incoming number and not the voip dial out  number?  Secondly, when the  second ring group kicks in only one of the external numbers dials when both  internal extensions and both external numbers should ring according to the ring  group setting. Any ideals what's going wrong?  Kind  regards  Brian.  UK 
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[Asterisk-Users] ISDN

[Mimmus]

Hi,
since last august, I worked on Asterisk and PRI lines, making a good
experience.
Now I need some information about ISDN BRI: I know that there is no native
support in Asterisk and I need some third-part driver. Then I read often
about different cards (Junghanss, Eicon, Beronet, etc), different drivers
(msidn, visdn, etc), instruction to patch Asterisk, etc
Could some goodwill man summarize this topic for me before I engage myself
in the rediscovery of warm water?

Thanks
-- 
Domenico Viggiani

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[Asterisk-Users] New to the list.

[Fran Serrano]

Hi all,

First of all I wanted to give my salutations to everyone in the list.

In second hand, let's see if anyone may help me with a problem i've got.

I bought a Developer Kit from Digium (TDM11B + ST2030 + Asterisk book)

I installed a new server over [EMAIL PROTECTED] with the card plugged in it.

Everything during the installation went fine.

Now I connected the telephone wire (from the telephone company) into
the FXO RJ11.

The problem is that, using a X-Lite softphone SIP client i'm not
able to call anyone from outside, I mean a regular telephone number.
It registers the extension OK.

When I try calling anyone, I always get a loopback connection,
hearing myself.

When I try from the ST2030 IP Hardware Phone I get two short tones
and that's it, and when I try to call extension 200 (the ST2030) from
extension 201 (Softphone) I always get a Digital receptionist telling me
that extension 200 is on the phone.

Could anyone give me a tip about where should be the problem? Is
there anything I need to dial before the number? How should I procced?

Any idea?


Thanks a lot in advance.

Fran.
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RE: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[shadowym]

That feature is called Bridged (or Shared) line appearance.  That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy.  There has been some talk about
implementing it but so far there does not seem to be any progress.

I was actually quite surprised to learn that it didn't do that. I am also
surprised that there doesn't seem to be much interest by developers in
general to make it do that even thought it is a deal killer for a lot of
people.

I would really like to see that feature myself as I have had to turn away
potential business because of the lack of that one feature.  If I could
write code I'd try do it myself. 

 -Original Message-
 From: Daniel Salama [mailto:[EMAIL PROTECTED] 
 Sent: Friday, June 23, 2006 2:00 AM
 To: Non-Commercial Discussion Asterisk
 Subject: [Asterisk-Users] GXP-2000 and Shared Line Appearances
 
 I have a client with 20 GXP-2000s. Everything seems to be 
 working fine. However, after a couple of weeks of use, the 
 client is having a hard time adjusting to the new IP based 
 phone systems and only misses one feature from their old 
 Lucent system.
 
 That is, they had 8 analog lines before and all their old 
 Lucent phones showed a button for each line. So, it was easy 
 for anyone to say, pick up line 2 or anyone to see which 
 lines were in use.
 
 Is it possible to use the GXP-2000 line buttons or extension 
 buttons to show the lines in use, shared by all phones. Since 
 the client is purchasing 8 virtual lines, I have them 
 restricted in a call group and also with incoming and 
 outgoing call limits. Is it possible for all the GXP-2000s to 
 show that line 1 is in use, and so on?
 
 Thanks,
 Daniel
 
 
 
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Re: [Asterisk-Users] call quality statistics?

[Andy Kuo]

try iax2 show netstats


On 6/23/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:

Is it possible to set up some sort of call-quality statistics
reporting/logging for IAX2 calls? Something that can keep track of
dropped packet / jitter trends?

(I know iax2 show channels shows this info for active calls.)

Suggestions appreciated!


- Mike

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Re: [Asterisk-Users] Asterisk Users Group - Portugal

[Josué Conti]

Marco, boa tarde (in Brazil)
Na realidade tenho interesse em estar participando deste sua idéia também, posso contribuir no que for necessário para que avance a idéia.

Cumprimentos 

Josué

I have interest inparticipating of this its idea , I can contribute in that will be necessary so that it advances the idea.

Best Regards.

2006/6/23, Marco Mouta [EMAIL PROTECTED]:
Boa tarde,Após alguma experiência com o Asterisk, e com muito ainda paraaprender, gostaria de saber se há alguém nesta mailing list que
pretenda criar um Asterisk Users Group para Portugal.Visto que acaba sempre por ser uma enorme aprendizagem ( valoracrescentado) a partilha de experiências/problemas e soluções nasimplementações Asterisk.
Há spre detalhes que variam entre os Telco's de cada país, voice prompts, etc.Se houver um número minimo de pessoas interessadas, podemos avançar com a ideia.--Com os melhores cumprimentos,
Marco Mouta___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] best hardphone for Asterisk?

[Andrei (MPI)]

My fellow employees like their Polycom 600s even more.

Andrei (MPI)

Jonathan k. Creasy wrote:

I'll second that. I really like the provisioning features. My customers
prefer the 501 because they like the layout and speaker phone
functionality. 


-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
West
Sent: Friday, June 23, 2006 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] best hardphone for Asterisk?

I find the Polycom Soundpoint 301 and 501 models to be great phones.

Christian Victor wrote:
  

Crazy Boy schrieb:
  


We have implemented Asterisk in our organization. There are 150
  

members in our organization. At present all are using softphones. Now, I
want to buy hardphones for our staff. Can anybody suggest me that what
is the best hardphone for Asterisk with low-cost?
  

  

I would say a Swissvoice IP 10S, a Snom 300 or - if you want better
quality - a Polycom 300.

The Snom looks good and is solid, the Swissvoice is similar plus it
supports PoE, the Polycom is a bit more expensive but worth the
additional cost.

Chris
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[Asterisk-Users] Re: Asterisk-1.2.9.1 e MOH

[Josué Conti]

But to register in the list, I compiled asterisk-1.2.9.1 with addons and he functioned. Thank'sJosué
2006/6/23, Josué Conti [EMAIL PROTECTED]:




Hi All
Somebody knows asresolv the error below? Already I compiled asterisk-addons-1.2.3, but exactly thus it reports this error, could help me? 



-- Executing WaitMusicOnHold(SIP/3205-d9ef, 30) in new stack -- Started music on hold, class 'default', on SIP/3205-d9efJun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 
Jun 23 02:14:21 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e4e45a8aJun 23 02:14:21 WARNING[24960]: 
layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 1758 bits!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303Jun 23 02:14:21 WARNING[24960]: common.c
 :134 decode_header: Layer 1 not supported!Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e4e45a8aJun 23 02:14:21 WARNING[24960]: layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 1758 bits! 
Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame Jun 23 02:14:21 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame 0b00eaffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame ebfff3ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 33003700Jun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 1f001800Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e2ffd8ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame dcffddffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame d6ffc6ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 1f002300Jun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame cfffc0ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 20001a00Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame f7ff1300Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e1ffe7ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame daffd4ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame 2a003d00Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame dbffd8ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame c6ffc5ffJun 23 02:14:22 WARNING[24960]: 
common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame ddfff1ffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! 
Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e3ffecffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: 
interface.c:215 decodeMP3: Junk at the beginning of frame d7ffecffJun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame d8ffd3ff 
Jun 23 02:14:22 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported!Jun 23 02:14:22 WARNING[24960]: interface.c:215 decodeMP3: Junk at the 

Re: [Asterisk-Users] ISDN

[Hermann Wecke]

Mimmus wrote:

Could some goodwill man summarize this topic for me before I engage
myself in the rediscovery of warm water?


Read a topic posted a few days ago: ISDN BRI NetJet
You will find good advice there.
If you need to buy a Cologne chipset card, check here:
http://www.solwise.co.uk/isdn.htm
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Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

[Dustin Wildes]

shadowym wrote:


That feature is called Bridged (or Shared) line appearance.  That is one of
the things Asterisk cannot do and nobody seems very interested in making it
do that because it is apparently not easy.  There has been some talk about
implementing it but so far there does not seem to be any progress.
 



http://forums.digium.com/viewtopic.php?p=23974#23974
I will be posting the code later today.


--Dustin Wildes
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RE: [Asterisk-Users] best hardphone for Asterisk?

[shadowym]

I love my Aastra 9133i with v1.4 firmware.  Pretty much everything just
works with Asterisk right out of the box and it has all the features I need.


 -Original Message-
 From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED] 
 Sent: Friday, June 23, 2006 8:03 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
 
 I'll second that. I really like the provisioning features. My 
 customers prefer the 501 because they like the layout and 
 speaker phone functionality. 
 
 -Jonathan
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joshua West
 Sent: Friday, June 23, 2006 10:20 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] best hardphone for Asterisk?
 
 I find the Polycom Soundpoint 301 and 501 models to be great phones.
 
 Christian Victor wrote:
  Crazy Boy schrieb:

  We have implemented Asterisk in our organization. There are 150
 members in our organization. At present all are using 
 softphones. Now, I want to buy hardphones for our staff. Can 
 anybody suggest me that what is the best hardphone for 
 Asterisk with low-cost?
  
 
  I would say a Swissvoice IP 10S, a Snom 300 or - if you want better 
  quality - a Polycom 300.
 
  The Snom looks good and is solid, the Swissvoice is similar plus it 
  supports PoE, the Polycom is a bit more expensive but worth the 
  additional cost.
 
  Chris
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 --
 Joshua West
 Linux Infrastructure Engineer
 Boston Engineering Corporation
 http://www.boston-engineering.com
 
 
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[Asterisk-Users] troubleshooting echo on speakerphone

[T. Shaw]

Hello all,
I'm looking to troubleshoot some echo issues and possibly studdering while 
using the speakerphone with the Intellitouch ITC3002. On the hardware end, i 
have a 2621 setup as the router with some policy maps for qos, and a 2900XL 
with cos priority set to 5. I have setup created 2 vlans currently (really 
small office), 1 for corp/data, and 2 voice vlan. Asterisk is on the 2nd 
vlan as well as the phones and i have the 2621 doing intervlan routing.


Currently we have VOIP calls coming in and out, but when calls are placed 
and speakerphone is utilized, the FAR end hears echo of themselves. This 
would happen regardless of who placed the call.


Any ideas on where i can start cutting down the echo problem?
The phones come with echo-cancellation and is turned on. Thephones also have 
VAD but is currently not on. Would this make a difference while utilizing 
the speakerphone?


Terrelle


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Re: [Asterisk-Users] Asterisk Users Group - Portugal

[Marco Mouta]

Olá  a todos! :)
Hi all,
Thanks for all of your fast replies, further now may be simpler to
talk in english. But any one could write in Portguese, any way the
important is to make it happen.

This weekend i will post here some topics could be interesting:


mailing list and a blog or php forum don't know, i must say i'm not a
web expert.

I work with VoiceXML VoIP more related to communications.

Mailing list and blog or forum seems easy to start this, share and learn.

I hope i can help to this project grow.

Best regards,
Marco Mouta




On 6/23/06, Josué Conti [EMAIL PROTECTED] wrote:


Marco, boa tarde (in Brazil)
Na realidade tenho interesse em estar participando deste sua idéia também,
posso contribuir no que for necessário para que avance a idéia.

Cumprimentos

Josué


I have interest in participating of this its idea , I can contribute in that
will be necessary so that it advances the idea.
Best Regards.


2006/6/23, Marco Mouta [EMAIL PROTECTED]:

Boa tarde,

Após alguma experiência com o Asterisk, e com muito ainda para
aprender, gostaria de saber se há alguém nesta mailing list que
pretenda criar um Asterisk Users Group para Portugal.

Visto que acaba sempre por ser uma enorme aprendizagem ( valor
acrescentado) a partilha de experiências/problemas e soluções nas
implementações Asterisk.

Há spre detalhes que variam entre os Telco's de cada país, voice prompts,
etc.


Se houver um número minimo de pessoas interessadas, podemos avançar com a
ideia.

--
Com os melhores cumprimentos,

Marco Mouta
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--
Com os melhores cumprimentos,

Marco Mouta
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RE: [Asterisk-Users] Re: Out of Office Auto Reply:

[Andrew Kirch]

I'd second this motion, this is very very annoying.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Colin Anderson
 Sent: Friday, June 23, 2006 11:20 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Re: Out of Office Auto Reply:
 
 Should be part of the FAQ for the list, as well as the setting for
 Exchange
 5.5 which a *lot* of orgs still run (we do too)
 
 I wonder if the list SW can be modded to automatically plonk any mail
with
 the subject string: Out of Office 
 
 
 -Original Message-
 From: Steven [mailto:[EMAIL PROTECTED]
 Sent: Friday, June 23, 2006 8:08 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Re: Out of Office Auto Reply:
 
 
 Exchange changes
 
 http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp
 
 --
 --
 Steven
 
 http://www.glimasoutheast.org
 
 
 
 Koopmann, Jan-Peter [EMAIL PROTECTED] wrote in
message

news:[EMAIL PROTECTED]
 On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote:
 
  We use MS Exchange too and, as far as I am aware, it is cognizant of
  mailing list headers and doesn't send OOO notices to mailing list
  postings. The only mailing list from which I receive my own OOO
  notices is one that doesn't have the proper mailing list headers
set.
 
 
 No. Exchange does not honour Precedence headers. It has some funky
way
 of
 determining what is a mailing list and what is not. It
 does not work very well and it has (or had) to be enabled via a
registry
 key. If you don't do this, even Exchange 2003 will reply to
 some mailing lists. But it should not send this to every mail but only
 once
 day...
 
 
 Kind regards,
   JP
 
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Re: [Asterisk-Users] best hardphone for Asterisk?

[Peter Antonacci]

The Polycom 501's or 601's are the way to go
On 6/23/06, shadowym [EMAIL PROTECTED] wrote:
I love my Aastra 9133i with v1.4 firmware.Pretty much everything justworks with Asterisk right out of the box and it has all the features I need.
 -Original Message- From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED]] Sent: Friday, June 23, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] best hardphone for Asterisk? I'll second that. I really like the provisioning features. My customers prefer the 501 because they like the layout and speaker phone functionality.
 -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Joshua West Sent: Friday, June 23, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] best hardphone for Asterisk?
 I find the Polycom Soundpoint 301 and 501 models to be great phones. Christian Victor wrote:  Crazy Boy schrieb:   We have implemented Asterisk in our organization. There are 150
 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost?
I would say a Swissvoice IP 10S, a Snom 300 or - if you want better  quality - a Polycom 300.   The Snom looks good and is solid, the Swissvoice is similar plus it
  supports PoE, the Polycom is a bit more expensive but worth the  additional cost.   Chris  ___  --Bandwidth and Colocation provided by 
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http://lists.digium.com/mailman/listinfo/asterisk-users  -- Joshua West Linux Infrastructure Engineer Boston Engineering Corporation 
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RE: [Asterisk-Users] best hardphone for Asterisk?

[Dave Cotton]

On Fri, 2006-06-23 at 10:39 -0700, shadowym wrote:
 I love my Aastra 9133i with v1.4 firmware.  Pretty much everything just
 works with Asterisk right out of the box and it has all the features I need.

If cost is important the 9112i would be better. I install all three
Aastra models and the sound quality is good across the range. 
-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] troubleshooting echo on speakerphone

[T. Shaw]

ok, ill try that as well. Funny thing is i don't have these issues using the 
same phone on my test box at home which is a old 256M Sun Ultra5 connected 
to a old Netgear Cable/DSL wireless router (can you say 0 (zero) QOS).


Terrelle





From: Colin Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'asterisk-users@lists.digium.com

Subject: RE: [Asterisk-Users] troubleshooting echo on speakerphone
Date: Fri, 23 Jun 2006 11:59:03 -0600

Sounds like a reflection echo i.e. the customer's voice is coming out of 
the

speaker, hitting a flat wall, and is being picked up by the mic on the
speakerphone. Try attenuating the microphone, or to test for it, move the
speakerphone to a smaller room or one with fabric walls like a cubicle.
Dollars to doughnuts, it goes away.

As to the stuttering, it is most likely because the audio is being
suppressed during speech or turned to half-duplex, which speakerphones are
notorious for. Speakerphones, voip or not, are tricky beasts and I try to
discourage their use amongst my users whenever possible. I personally
dislike speakerphones; I consider them rude.

-Original Message-
From: T. Shaw [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 11:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] troubleshooting echo on speakerphone


Hello all,
I'm looking to troubleshoot some echo issues and possibly studdering 
while


using the speakerphone with the Intellitouch ITC3002. On the hardware end, 
i


have a 2621 setup as the router with some policy maps for qos, and a 2900XL
with cos priority set to 5. I have setup created 2 vlans currently (really
small office), 1 for corp/data, and 2 voice vlan. Asterisk is on the 2nd
vlan as well as the phones and i have the 2621 doing intervlan routing.

Currently we have VOIP calls coming in and out, but when calls are placed
and speakerphone is utilized, the FAR end hears echo of themselves. This
would happen regardless of who placed the call.

Any ideas on where i can start cutting down the echo problem?
The phones come with echo-cancellation and is turned on. Thephones also 
have


VAD but is currently not on. Would this make a difference while utilizing
the speakerphone?

Terrelle


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[Asterisk-Users] Asterisk home on VMWare time sync issues

[Al Lougher]

Hello - I am using AH on VMWARE. I have noticed that the date and time periodically loses sync with the server system time. Does anyone know why this would be, or where if I need to change a setting somewhere so it keeps time properly?Thanks!  Alan. 
		Want to be your own boss? Learn how on  Yahoo! Small Business. 
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[Asterisk-Users] Odd SIP error message

[Johann]

As of late, I keep seeing a very odd error message in Asterisk and regardless of
how much debugging or verbose I set I can't get more detailed info to find out
what exactly is causing the error.  It's every few seconds and in no regular
pattern either.

Jun 23 05:24:17 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:18 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:20 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:24 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:28 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:32 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:36 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:40 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum
Jun 23 05:24:44 NOTICE[29057] chan_sip.c: SIP: Received packet with bad UDP 
checksum

Any advice on what I can do to get more information?  I see it occur when
Asterisk doesn't think there are any active channels going on as well.  There's
a SIP device that is misbehaving somewhere.  Most of the SIP phones are Polycom
IP600/601s.


--johann
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[Asterisk-Users] QueueMetrics 1.2 released today

[lenz]

Hello list,

I am pleased to tell you that we have released a new version of  
QueueMetrics. The main areas of improvement were the following ones:


- Unattended client monitoring: for large call centers who work for third  
parties, it is possible to have your clients log in directly to  
QueueMetrics, see a reduced real-time page and monitor calls and agents in  
realtime.
- VNC monitoring: it is possible for supervisors and clients to launch a  
third-party VNC screen monitoring app to see the screen of your agents in  
real-time
- Audio monitoring: it is possible for supervisors and clients to click on  
an ongoing call and listen to it through their telephone
- New feature: report mode. It is now possible to download both the  
current real-time page and the current full analysis using a simple URL  
(useful for automating report extraction using a cron job).

- New feature: inclusive SLA - computes both taken and lost calls.
- New feature for [EMAIL PROTECTED]/TrixBox users: all channels in the format  
Local/[EMAIL PROTECTED] can be rewritten as Agent entries.
- Explicit session timeout set in web.xml - easier for users to change as  
needed.
- QM is now able to correctly parse [EMAIL PROTECTED]/TrixBox generated  
configuration files.


A full list of improvements over version 1.1 can be found at  
http://queuemetrics.loway.it/news.jsp


QueueMetrics 1.2 allows data storage on both flat files and MySQL  
databases for bigger call centers. And of course comes with a 90-page user  
manual that covers all aspects of it.


QueueMetrics is a commercial call center monitoring package, but is  
availabe free of charge for individuals, Asterisk hackers and small SOHOs.  
You can request a trial key if you run a larger installation and would   
like to test it in your own environment.


The latest version of QueueMetrics can be downloaded from  
http://queuemetrics.loway.it/download.jsp


Hope you like it,
l.


--
Assum est, versa et manduca.
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RE: [Asterisk-Users] troubleshooting echo on speakerphone

[Colin Anderson]

Different room. Different acoustic characteristics. 

-Original Message-
From: T. Shaw [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 12:21 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] troubleshooting echo on speakerphone


ok, ill try that as well. Funny thing is i don't have these issues using the

same phone on my test box at home which is a old 256M Sun Ultra5 connected 
to a old Netgear Cable/DSL wireless router (can you say 0 (zero) QOS).

Terrelle




From: Colin Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] troubleshooting echo on speakerphone
Date: Fri, 23 Jun 2006 11:59:03 -0600

Sounds like a reflection echo i.e. the customer's voice is coming out of 
the
speaker, hitting a flat wall, and is being picked up by the mic on the
speakerphone. Try attenuating the microphone, or to test for it, move the
speakerphone to a smaller room or one with fabric walls like a cubicle.
Dollars to doughnuts, it goes away.

As to the stuttering, it is most likely because the audio is being
suppressed during speech or turned to half-duplex, which speakerphones are
notorious for. Speakerphones, voip or not, are tricky beasts and I try to
discourage their use amongst my users whenever possible. I personally
dislike speakerphones; I consider them rude.

-Original Message-
From: T. Shaw [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 11:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] troubleshooting echo on speakerphone


Hello all,
I'm looking to troubleshoot some echo issues and possibly studdering 
while

using the speakerphone with the Intellitouch ITC3002. On the hardware end, 
i

have a 2621 setup as the router with some policy maps for qos, and a 2900XL
with cos priority set to 5. I have setup created 2 vlans currently (really
small office), 1 for corp/data, and 2 voice vlan. Asterisk is on the 2nd
vlan as well as the phones and i have the 2621 doing intervlan routing.

Currently we have VOIP calls coming in and out, but when calls are placed
and speakerphone is utilized, the FAR end hears echo of themselves. This
would happen regardless of who placed the call.

Any ideas on where i can start cutting down the echo problem?
The phones come with echo-cancellation and is turned on. Thephones also 
have

VAD but is currently not on. Would this make a difference while utilizing
the speakerphone?

Terrelle


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RES: [Asterisk-Users] Meetme max users

[cleviton.araujo]

Hi, Matt:

What´s your server specifications that did you use?

Best Regards,
Cleviton.


-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nome de Matt Florell
Enviada em: sexta-feira, 23 de junho de 2006 11:38
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [Asterisk-Users] Meetme max users


We've had over 100 participants spread across 30 meetme rooms on a
single server before,  and the most we've had in a single meetme room
is 46. I don't know of a hard limit for meetme participants and I
haven't seen a limit in the code. You would most likely be limited by
the resources on your server I would guess.

MATT---

On 6/23/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote:
 Does anyone knows what is the max of users that meefme can handle.
 I am using Iax2 clients to connect to the conference.

 Thanks


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[Asterisk-Users] Voice calls sent to fax extension

[Paul A. Pringle]

I have a situation that has repeated itself a few times.  Someone calls
into Asterisk and is connected with a voice extension.  At some point
during the call, the log shows chan_zap.c: DTMF digit: f on Zap/2-1.
At this point, the call is redirected to receive a fax and the Asterisk
voice extension is hung up.  The users report that there were no
noticable tones heard just before the cutover, so I'm not sure what's
going on.  Is there a way to disable detection of faxes after the
voicecall is initiated?  We're running a Digium card to convert our
analog trunks if that makes any difference.

Thanks!

Paul
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