Re: [asterisk-users] Mexico Dids

[Gideon Hack]

Hi Robert,

DID World Wide has coverage for Saltillo (please see 
http://www.didww.com/virtual_numbers/Mexico), with flat-rate forwarding to 
PSTN, VoIP, SIP, H.323, IAX, Skype, MSN or Google Talk. 


Regards,
Gideon

From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 22 Feb 2008 18:03:00 -0500
Subject: [asterisk-users] Mexico Dids








Hi,
I am looking 
for a did from Saltillo 
Mexico.
Any pointers?
robert

_

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Re: [asterisk-users] Music on hold

[Fons van der Beek]

I've overwritten the indications.conf with the one from the sourcecode, 
stil no luck
Perhaps somebody knows what the correct value for indications.conf is 
when using the dutch xs4all as sip carrier??


and even with verbose set to 114 (quite big) there are no errormessages 
indicating that something is wrong with indications (in respect to syntax)


Eric Wieling schreef:
If not answering fixes the problem then the issue is indications.conf. 
Try using the indications.conf.sample file included with the Asterisk 
source code, then stop Asterisk and starting it again.  I do not know if 
indications.conf is reloaded on a reload.


Fons van der Beek wrote:
  

NOT answering did the trick!
Tnx a lot! now it works like it should work!



Eric Wieling schreef:

Replying to my own post.  Asterisk uses indications.conf when it has 
to provide tones AFTER the line is answered.  You might get a message 
on the console like "Unable to handle indication 15" or something like 
that.


Eric Wieling wrote:
 
  
Don't answer the line.  Also try using the US indications, just in 
case something odd is in the NL setup.


Fons van der Beek wrote:
   


Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an 
external line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:
 
  
This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 
   


I tried that, its gives me the same problem.

Kevin P. Fleming schreef:

  

Fons van der Beek wrote:
  

Because i want a ringing signal while people are in a waiting 
queue i've created a wav file containing our local ringing 
indication
If I make an inside call to the queue, the correct sound is 
played, but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok
  
  
The Queue() application has an option to generate ringback to 
callers
instead of music on hold, why don't you just use that instead of 
trying

to craft a new solution?




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Re: [asterisk-users] FXO Cards - T38

[Steve Underwood]

Rob Hillis wrote:
> Not unless you're running CallWeaver or Asterisk 1.6.0-beta4.  Asterisk 
> has had passthrough support for T.38 for a while (somewhere in 1.4 it 
> became available IIRC) but is currently completely incapable of 
> terminating or encoding a fax call to T.38.
>   
I thought * was still not capable for T.38 gateway operation. Doesn't 
beta 4 just added T.38 termination? And, I believe it misses out some 
key elements of doing that properly. Note that T.38 termination is an 
addon, so it can't be used with, say, G.729.
> The only real option available at the moment is to keep one PSTN line on 
> an ATA with an FXO port and T.38 support available and direct calls from 
> the fax machines through to it.  However, I should point out that while 
> I believe this should be possible, I haven't actually tried it myself.
>
>   
Steve


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Re: [asterisk-users] AGI / Voicemail Que

[Trevor Peirce]

Nitesh Divecha wrote:
> Everything is working fine but the only problem is voice mail greetings 
> for Busy and Unavailable is not played. By default only "Temp Greetings" 
> voice mail greetings is played. I am passing the correct parameters for 
> Busy => 'b', Unavailable => 'u' and default goes to "Not Answered".
>   
I believe the temp greeting will override busy and unavailable 
greetings. Delete the temp greeting and you should have no problem.

Trevor

-- 
Real CNAM data for incoming Caller ID @ www.cnam.info


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[asterisk-users] Monitor Asterisk

[Michael Henderson]

Hi, I have some experience with Asterisk. What I would like to know is, are
there any programmable APIs that we can use to get the information monitored
by asterisk.
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Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[Yehavi Bourvine +972-8-9489444]

Hello,

> I've one nokia E65 that works very well with my asterisk box.

The people here don't let me even try it as they are afraid it will consume the
battery more than when it is used "the usual way". Is this true?

  Thanks, __Yehavi:

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Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[Sanspareils Greenlands]

I have an E61i and it works great with my Asterisk. No
extra software needed, everything is built into those phones. 

Rajeev.

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Re: [asterisk-users] AGI / Voicemail Que

[Nitesh Divecha]

Thanks Doug,

I tried that but it didn't work either... As per Wiki 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it 
has a statement that starting from 1.4-trunk FLAG must be pass using a 
pipe sign '|'.

I have other Asterisk 1.2 running with FreePBX and I went over the agi 
code, I saw its passing FLAG using pipe sign '|'.

So now I am kinda confused...

Cheers,
Nitesh





Doug Lytle wrote:
> Nitesh Divecha wrote:
>   
>> ([EMAIL PROTECTED]|b)
>>
>> Any suggestions... By the way I am running Asterisk 1.2.18
>>
>>   
>> 
>
> I believe under 1.2.x it would be [EMAIL PROTECTED]
>
> One of my older dial plans lists:
>
> s-BUSY,1,Voicemail([EMAIL PROTECTED])
>
> Doug
>
>
>   


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Re: [asterisk-users] load balancing SIP extensions

[Jared Bellows]

>
> I tried to use DUNDi on my local servers but I can't
> seem to make it work. Most howtos out there explain
> the use of DUNDi when the extension ranges do not
> overlap.
>

The following doc describes using the same extensions across multiple *
servers. It requires using realtime, but seems to do what you describe.

http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf
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Re: [asterisk-users] Music on hold

[Eric Wieling]

If not answering fixes the problem then the issue is indications.conf. 
Try using the indications.conf.sample file included with the Asterisk 
source code, then stop Asterisk and starting it again.  I do not know if 
indications.conf is reloaded on a reload.

Fons van der Beek wrote:
> NOT answering did the trick!
> Tnx a lot! now it works like it should work!
> 
> 
> 
> Eric Wieling schreef:
>> Replying to my own post.  Asterisk uses indications.conf when it has 
>> to provide tones AFTER the line is answered.  You might get a message 
>> on the console like "Unable to handle indication 15" or something like 
>> that.
>>
>> Eric Wieling wrote:
>>  
>>> Don't answer the line.  Also try using the US indications, just in 
>>> case something odd is in the NL setup.
>>>
>>> Fons van der Beek wrote:
>>>
 Tnx.

 I checked /etc/asterisk/indications.conf and my default location nl
 is listed in the options

 So i am still puzzled


 my extensions.conf in respect to incomming calls (as basic as possible)

 exten => s,1,Answer
 exten => s,2,queue(receptie|r)
 exten => s,3,Voicemail(201)

 everything else works as it should work, but no "ringing" on an 
 external line

 on the other hand, internaly: it's ok

 exten => 205,1,queue(receptie|r)
 exten => 205,2,busy

 205 gives ringing


 Eric Wieling schreef:
  
> This problem would happen if you did not have 
> /etc/asterisk/indications.conf
>
> Fons van der Beek wrote:
>  
>
>> I tried that, its gives me the same problem.
>>
>> Kevin P. Fleming schreef:
>> 
>>> Fons van der Beek wrote:
>>>   
 Because i want a ringing signal while people are in a waiting 
 queue i've created a wav file containing our local ringing 
 indication
 If I make an inside call to the queue, the correct sound is 
 played, but when i make an external call, no signal is heard.
 everything else looks ok, and all other functions are ok
   
>>> The Queue() application has an option to generate ringback to 
>>> callers
>>> instead of music on hold, why don't you just use that instead of 
>>> trying
>>> to craft a new solution?
>>>
>>> 
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>>   
>   
  


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>>
>>   
> 
> 
> 
> 
> 
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Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors

[Mike Hammett]

It was my understanding that voicemail.conf referenced MySQL and not 
asterisk.


--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, February 22, 2008 6:56 PM
Subject: Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors


> On Friday 22 February 2008 18:28:56 Mike Hammett wrote:
> --snip--
>> [asterisk]
>> enabled => no
>> dsn => asterisk
>> ;username => myuser
>> ;password => mypass
>> pre-connect => yes
>
> --snip--
>> WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain 
>> database
>> object for 'asterisk'!
>
> What does "enabled" mean to you?
>
> -- 
> Tilghman
>
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

NOT answering did the trick!
Tnx a lot! now it works like it should work!



Eric Wieling schreef:
Replying to my own post.  Asterisk uses indications.conf when it has to 
provide tones AFTER the line is answered.  You might get a message on 
the console like "Unable to handle indication 15" or something like that.


Eric Wieling wrote:
  
Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.


Fons van der Beek wrote:


Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:
  
This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 


I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
   
  

Fons van der Beek wrote:
   

Because i want a ringing signal while people are in a waiting queue 
i've created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, 
but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok

  

The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?




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Re: [asterisk-users] Music on hold

[Fons van der Beek]

it's very odd
-I just upgraded to 1.4.18 (from 1.4.17)
-removed answer
-changed to several other options, still no luck
(restarted also)





Eric Wieling schreef:
Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.


Fons van der Beek wrote:
  

Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:

This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 
  

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
   


Fons van der Beek wrote:
   
  
Because i want a ringing signal while people are in a waiting queue 
i've created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, 
but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok



The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?


  

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Re: [asterisk-users] Music on hold

[Eric Wieling]

Replying to my own post.  Asterisk uses indications.conf when it has to 
provide tones AFTER the line is answered.  You might get a message on 
the console like "Unable to handle indication 15" or something like that.

Eric Wieling wrote:
> Don't answer the line.  Also try using the US indications, just in case 
> something odd is in the NL setup.
> 
> Fons van der Beek wrote:
>> Tnx.
>>
>> I checked /etc/asterisk/indications.conf and my default location nl
>> is listed in the options
>>
>> So i am still puzzled
>>
>>
>> my extensions.conf in respect to incomming calls (as basic as possible)
>>
>> exten => s,1,Answer
>> exten => s,2,queue(receptie|r)
>> exten => s,3,Voicemail(201)
>>
>> everything else works as it should work, but no "ringing" on an external 
>> line
>>
>> on the other hand, internaly: it's ok
>>
>> exten => 205,1,queue(receptie|r)
>> exten => 205,2,busy
>>
>> 205 gives ringing
>>
>>
>> Eric Wieling schreef:
>>> This problem would happen if you did not have 
>>> /etc/asterisk/indications.conf
>>>
>>> Fons van der Beek wrote:
>>>  
 I tried that, its gives me the same problem.

 Kevin P. Fleming schreef:

> Fons van der Beek wrote:
>
>> Because i want a ringing signal while people are in a waiting queue 
>> i've created a wav file containing our local ringing indication
>> If I make an inside call to the queue, the correct sound is played, 
>> but when i make an external call, no signal is heard.
>> everything else looks ok, and all other functions are ok
>> 
> The Queue() application has an option to generate ringback to callers
> instead of music on hold, why don't you just use that instead of trying
> to craft a new solution?
>
> 
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>>>   
>>
>>
>> 
>>
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> 

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Music on hold

[Eric Wieling]

Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.

Fons van der Beek wrote:
> Tnx.
> 
> I checked /etc/asterisk/indications.conf and my default location nl
> is listed in the options
> 
> So i am still puzzled
> 
> 
> my extensions.conf in respect to incomming calls (as basic as possible)
> 
> exten => s,1,Answer
> exten => s,2,queue(receptie|r)
> exten => s,3,Voicemail(201)
> 
> everything else works as it should work, but no "ringing" on an external 
> line
> 
> on the other hand, internaly: it's ok
> 
> exten => 205,1,queue(receptie|r)
> exten => 205,2,busy
> 
> 205 gives ringing
> 
> 
> Eric Wieling schreef:
>> This problem would happen if you did not have 
>> /etc/asterisk/indications.conf
>>
>> Fons van der Beek wrote:
>>  
>>> I tried that, its gives me the same problem.
>>>
>>> Kevin P. Fleming schreef:
>>>
 Fons van der Beek wrote:

> Because i want a ringing signal while people are in a waiting queue 
> i've created a wav file containing our local ringing indication
> If I make an inside call to the queue, the correct sound is played, 
> but when i make an external call, no signal is heard.
> everything else looks ok, and all other functions are ok
> 
 The Queue() application has an option to generate ringback to callers
 instead of music on hold, why don't you just use that instead of trying
 to craft a new solution?

 
>>> ___
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>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>> 
>>
>>   
> 
> 
> 
> 
> 
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T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors

[Tilghman Lesher]

On Friday 22 February 2008 18:28:56 Mike Hammett wrote:
--snip--
> [asterisk]
> enabled => no
> dsn => asterisk
> ;username => myuser
> ;password => mypass
> pre-connect => yes

--snip--
> WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database
> object for 'asterisk'!

What does "enabled" mean to you?

-- 
Tilghman

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[asterisk-users] MySQL Voicemail Storage Questions\Errors

[Mike Hammett]

I am running CentOS 5 with Asterisk 1.4.14.  I am trying to setup storage of 
voicemail messages into MySQL.  It is my understanding that I can only do this 
via ODBC.  I installed per 
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation  
unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and 
mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to 
turn on ODBC voicemail storage.  Here is the output of some config files:

[EMAIL PROTECTED] asterisk]# cat /etc/odbcinst.ini
# Example driver definitinions
#
#

# Included in the unixODBC package
#[PostgreSQL]
#Description= ODBC for PostgreSQL
#Driver = /usr/lib/libodbcpsql.so
#Setup  = /usr/lib/libodbcpsqlS.so
#FileUsage  = 1


# Driver from the MyODBC package
# Setup from the unixODBC package
[MySQL]
Description = ODBC for MySQL
Driver  = /usr/lib64/libmyodbc3.so
Setup   = /usr/lib64/libodbcmyS.so
FileUsage   = 1
You have new mail in /var/spool/mail/root
[EMAIL PROTECTED] asterisk]# cat /etc/odbc.ini
[astrealtime]
Description = Asterisk realtime FUNC_ODBC access
Driver  = MySQL
Socket  = /var/lib/mysql/mysql.sock
Server  = localhost
User= astrealtime
Pass= 
Database= asterisk
Option  = 3
[EMAIL PROTECTED] asterisk]# cat  /etc/asterisk/res_odbc.conf
;;; odbc setup file

; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
; so you can't have different values for different connections.
[ENV]
INFORMIXSERVER => my_special_database
INFORMIXDIR => /opt/informix

; All other sections are arbitrary names for database connections.

[asterisk]
enabled => no
dsn => asterisk
;username => myuser
;password => mypass
pre-connect => yes


[mysql]
enabled => yes
dsn => MySQL-asterisk
username => astrealtime
password => 
pre-connect => yes

; Certain servers, such as MS SQL Server and Sybase use the TDS protocol, which
; limits the number of active queries per connection to 1.  By setting up pools
; of connections, Asterisk can be made to work with these servers.
[sqlserver]
enabled => no
dsn => mickeysoft
pooling => yes
limit => 5
username => oscar
password => thegrouch
pre-connect => yes



[EMAIL PROTECTED] asterisk]# cat /etc/asterisk/voicemail.conf
odbcstorage=mysql
odbctable=voicemail_messages

[EMAIL PROTECTED] asterisk]# asterisk -vvv | grep odbc
  == Parsing '/etc/asterisk/res_odbc.conf': Found
[Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:229 load_odbc_config: Adding ENV 
var: INFORMIXSERVER=my_special_database
[Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:229 load_odbc_config: Adding ENV 
var: INFORMIXDIR=/opt/informix
[Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:508 odbc_obj_connect: Connecting 
mysql
[Feb 22 18:21:46] WARNING[21214]: res_odbc.c:519 odbc_obj_connect: res_odbc: 
Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not 
found, and no default driver specified
[Feb 22 18:21:46] WARNING[21214]: res_odbc.c:444 ast_odbc_request_obj: Failed 
to connect to mysql
[Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:302 load_odbc_config: Registered 
ODBC class 'mysql' dsn->[MySQL-asterisk]
[Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:684 load_module: res_odbc loaded.
res_odbc.so => (ODBC Resource)
[Feb 22 18:21:46] NOTICE[21214]: config.c:1250 ast_config_engine_register: 
Registered Config Engine odbc
res_config_odbc loaded.
res_config_odbc.so => (ODBC Configuration)
  == Parsing '/etc/asterisk/func_odbc.conf': Found
func_odbc.so => (ODBC lookups)
  == Parsing '/etc/asterisk/cdr_odbc.conf': Found
cdr_odbc.so => (ODBC CDR Backend)



[EMAIL PROTECTED] asterisk]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 22 to server version: 5.0.22

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql> use asterisk;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Database changed
mysql> describe voicemail_messages;
++-+--+-+-++
| Field  | Type| Null | Key | Default | Extra  |
++-+--+-+-++
| id | int(11) | NO   | PRI | NULL| auto_increment |
| msgnum | int(11) | NO   | | 0   ||
| dir| varchar(80) | YES  | MUL | NULL||
| context| varchar(80) | YES  | | NULL||
| macrocontext   | varchar(80) | YES  | | NULL||
| callerid   | varchar(40) | YES  | | NULL||
| origtime   | varchar(40) | YES  | | NULL||
| duration   | varchar(20) | YES  | | NULL||
| mailboxuser| varchar(80) | YES  |  

Re: [asterisk-users] Music on hold

[Fons van der Beek]

Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:

This problem would happen if you did not have /etc/asterisk/indications.conf

Fons van der Beek wrote:
  

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:


Fons van der Beek wrote:
  
  
Because i want a ringing signal while people are in a waiting queue i've 
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but 
when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok



The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?

  
  

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Re: [asterisk-users] FXO Cards - T38

[Rob Hillis]

Not unless you're running CallWeaver or Asterisk 1.6.0-beta4.  Asterisk 
has had passthrough support for T.38 for a while (somewhere in 1.4 it 
became available IIRC) but is currently completely incapable of 
terminating or encoding a fax call to T.38.

The only real option available at the moment is to keep one PSTN line on 
an ATA with an FXO port and T.38 support available and direct calls from 
the fax machines through to it.  However, I should point out that while 
I believe this should be possible, I haven't actually tried it myself.


Fernando Berretta wrote:
> Hi,
>
> Could some one let me know if a fax is received through a FXO card 
> connected to * and fax machine is connected to a Linksys FXS device 
> which support T38, is T38 going to be used for faxes which comes from 
> PSTN or go through PSTN ? or because of Asterisk T38 passthrough support 
> it is not possible ? so is for this scenery better to use external FXO 
> gateways with t38 support ?
>
> Regards,
> Fernando
>
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Re: [asterisk-users] Music on hold

[Eric Wieling]

This problem would happen if you did not have /etc/asterisk/indications.conf

Fons van der Beek wrote:
> I tried that, its gives me the same problem.
> 
> Kevin P. Fleming schreef:
>> Fons van der Beek wrote:
>>   
>>> Because i want a ringing signal while people are in a waiting queue i've 
>>> created a wav file containing our local ringing indication
>>> If I make an inside call to the queue, the correct sound is played, but 
>>> when i make an external call, no signal is heard.
>>> everything else looks ok, and all other functions are ok
>>> 
>> The Queue() application has an option to generate ringback to callers
>> instead of music on hold, why don't you just use that instead of trying
>> to craft a new solution?
>>
>>   
> 
> 
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> 

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Linksys SPA-942 Phones

[Rob Hillis]

So far I've never run into anything that's even /close/ to the 
speakerphone quality of the Polycoms.  There's no comparison on the 
speakerphone  between the Linksys phones and the Polycoms - it's chalk 
and cheese, but by the same token that holds true for just about every 
other phone too.



Chris Bagnall wrote:

*   How is the speaker phone quality?



Excellent. Noticably better than either the Snom 370 or Snom 300 that also 
populate my desk. Polycoms are a bit harder to come by here in the UK, so I 
haven't had much experience with them and can't do a direct comparison, but 
definitely much better than the speakerphones in the Snom units. I would say on 
a par with the Cisco 7960 (which also graces my desk).

  
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[asterisk-users] Mexico Dids

[Robert Augustyn]

Hi,
I am looking for a did from Saltillo Mexico.
Any pointers?
robert
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Re: [asterisk-users] Polycom 301/501 Keymapping

[Mojo with Horan & Company, LLC]

That can be found in the monstrous admin guide for the phone, seemly in 
Section 3.1.7 in my ancient version 1.5.0 document.  It shows me that on 
the 501, that button is 9 instead of 23. 

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html

There's a link to the administrator's guide down there under Setup & 
Maintenance Documents.

Moj


Rob Schall wrote:
> I know how to remap a key on a polycom 301 and 501
>
> But does anyone know of a list of mapping keys?
>
> For example, the Do Not Disturb on a 301 is #23. I got that one by just
> guessing though.
>
> Thanks,
> Rob
>
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Re: [asterisk-users] Post call QoS in Asterisk 1.4

[[EMAIL PROTECTED]]

I have absolutely no idea since I was not even aware of it. However,  
this may give you some hints as to where you can find more information:


http://www.mail-archive.com/[EMAIL PROTECTED]/msg27124.html

- Waldo

On Feb 22, 2008, at 5:08 PM, Douglas Garstang wrote:

It's time to ask this question again. Maybe I will get a reply one  
day. :)


Asterisk 1.4 has some channel variables that you can inspect after a  
call is complete that will give you QoS metrics. Stuff like average  
round trip time, etc.


Since there's only one set of variables, and calls will have two  
channels, which channel is this information for? Is it for one of  
the channels? Is it an aggregate of both channels? Who added this  
code and what where they thinking when they wrote it?


Thanks,
Doug.

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Re: [asterisk-users] spandsp/tx_fax/rx_fax frustrations

[Jim Duda]

Edwin,

I feel your pain.  I struggled getting fax to work reliably with both 
1.2 and 1.4 versions.  Any combination I tried, usually caused a crash.

I recently upgraded to 1.6.beta4 and installed the app_fax from the 
addons installation and it worked first time out of the box :-)

I've received a couple dozen test faxes without any crashes.

I realize it doesn't help your 1.2 needs, but I can confirm your 
frustration.

Jim

Edwin Lam wrote:
> hi
> 
> does any body know which version combination of
> spandsp/tx_fax/rx_fax will work with * 1.2.24?
> 
> i tried different combo. they're either seg fault
> during runtime or won't compile.
> 
> very frustrated :/
> 
> p.s. i know. hylafax/iaxmodem is far more stable. but i have
> specific reasons to use rx_fax.
> 


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Re: [asterisk-users] Music on hold

[Fons van der Beek]

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
> Fons van der Beek wrote:
>   
>> Because i want a ringing signal while people are in a waiting queue i've 
>> created a wav file containing our local ringing indication
>> If I make an inside call to the queue, the correct sound is played, but 
>> when i make an external call, no signal is heard.
>> everything else looks ok, and all other functions are ok
>> 
>
> The Queue() application has an option to generate ringback to callers
> instead of music on hold, why don't you just use that instead of trying
> to craft a new solution?
>
>   


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Re: [asterisk-users] Music on hold

[Kevin P. Fleming]

Fons van der Beek wrote:
> Because i want a ringing signal while people are in a waiting queue i've 
> created a wav file containing our local ringing indication
> If I make an inside call to the queue, the correct sound is played, but 
> when i make an external call, no signal is heard.
> everything else looks ok, and all other functions are ok

The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] 1.4 and IAX Trunks ...

[Kevin P. Fleming]

Gordon Henderson wrote:

> What about the need for 1.4 at all sites? Is it sufficient to just have it 
> in the "man in the middle" site?

It uses new IAX2 commands, so it requires that all three endpoints
understand those commands. At this time the only IAX2 implementations
that I am aware of that support media-only transfers are Asterisk 1.4
and Asterisk 1.6 beta releases.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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[asterisk-users] Post call QoS in Asterisk 1.4

[Douglas Garstang]

It's time to ask this question again. Maybe I will get a reply one day. :)

Asterisk
1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip
time, etc.

Since there's only one set of variables, and calls will
have two channels, which channel is this information for? Is it for one
of the channels? Is it an aggregate of both channels? Who added this
code and what where they thinking when they wrote it?

Thanks,
Doug.



  

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Re: [asterisk-users] Digium B410P and 8 ports connectivity

[Kevin P. Fleming]

Olivier wrote:

> Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI
> ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be
> made out of 2 B410P ?

No, the card does not support that mode.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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[asterisk-users] Polycom 301/501 Keymapping

[Rob Schall]

I know how to remap a key on a polycom 301 and 501

But does anyone know of a list of mapping keys?

For example, the Do Not Disturb on a 301 is #23. I got that one by just
guessing though.

Thanks,
Rob

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Re: [asterisk-users] cid_rewrite.php -- Caller ID Name lookup

[Jay Milk]

Jay Milk wrote:
> For those folks who are still using it --
>
> I updated the cid_rewrite script.  I noticed that two of the providers 
> were "iffy" and one had changed format a little while ago.  It's working 
> again.
>
> http://muware.com/asterisk has the latest (1.2.0)
>   
Updated to 1.2.1 to fix an issue with how test_cid.php received 
parameters.  Thanks to the friendly soul for pointing this out.

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Re: [asterisk-users] Message waiting light on polycom 301 using asterisk 1.4.14

[Doug Lytle]

Cavanna, Richard wrote:
> All,
>
>   I am setting up asterisk on a nslu2 (Linksys) using unslug.
> Everything is working great except that I have a polycom 301 and I
> cannot get the message indicator to work. I have created the users and
> mailbox in users.conf and I can manually dial the mailbox (*986000).
> Last thing is I am not using config files for the polycom just web
> browser. 
>
> Can anyone point me in the right direction 
>
>
> I see this on the console  
>
> chan_sip.c:14907 handle_request_subscribe: Received SIP subscribe for
> peer without mailbox: 6000
>
>
> users.conf
>
> [6000]
>
>   
You're missing:

[EMAIL PROTECTED]

You'll also need to edit the voicemail.conf

Doug


-- 
 
Ben Franklin quote:

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



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Re: [asterisk-users] Linksys SPA-942 Phones

[Chris Bagnall]

> *   How do the phones handling system wide paging? Is it similar to the
> Polycom phones?

No idea I'm afraid, none of our clients use paging functionality.

> *   Can a corporate directory be configured with the phones using 
> Asterisk?

Yes and no. You can set up a directory on a per-phone basis through the web 
interface, but it isn't overwritten by a provisioning config. You can fudge the 
issue some with curl and a php (or insert choice of preferred scripting 
language) script.

> *   How is the speaker phone quality?

Excellent. Noticably better than either the Snom 370 or Snom 300 that also 
populate my desk. Polycoms are a bit harder to come by here in the UK, so I 
haven't had much experience with them and can't do a direct comparison, but 
definitely much better than the speakerphones in the Snom units. I would say on 
a par with the Cisco 7960 (which also graces my desk).

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons




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[asterisk-users] Message waiting light on polycom 301 using asterisk 1.4.14

[Cavanna, Richard]

All,

I am setting up asterisk on a nslu2 (Linksys) using unslug.
Everything is working great except that I have a polycom 301 and I
cannot get the message indicator to work. I have created the users and
mailbox in users.conf and I can manually dial the mailbox (*986000).
Last thing is I am not using config files for the polycom just web
browser. 

Can anyone point me in the right direction 


I see this on the console  

chan_sip.c:14907 handle_request_subscribe: Received SIP subscribe for
peer without mailbox: 6000


users.conf

[6000]
fullname = Joe User
email = [EMAIL PROTECTED]
secret = 1234
;zapchan = 1
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
context = default
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes


Thanks for any help you can provide.


Richard 


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[asterisk-users] is tos=ef same as tos=0xb8 same as DSCP ef ?

[sean darcy]

Trying to figure out how to prefer voip traffic on a dsl line.

Found a great howto:
http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk

but I'm trying to figure out the relationship between the tos of 
iax.conf and tos of tc from Iproute2. my traffic goes from my linux 
router to a CPE cisco box. I understand Cisco uses tos ( usually 
referred to as DSCP, just to keep us on our toes) ef or 0xb8.

So when I use tos=ef in *.conf, will the Cisco router see it as Cisco's 
"ef", DSCP=ef?

And, for extra credit, can I use tos 0xb8 in iproute2:
tc filter add dev eth1 protocol ip parent 1: prio 1 u32 match ip tos 
0xb8 0xff flowid 1:2

It looks to me that priomap only has 16 values, so anything over 0x10 
won't work.

Thanks

sean


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Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[Gordon Henderson]

On Sat, 23 Feb 2008, amit salunkhe wrote:

> Hi
>i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
> client so i can make VOIP calls thru that phone. Aslo that Phone easly able
> to register with Asterisk Pbx to recive inter-office calls.
> i try to search from web & also from Nokia site but they only mention this
> features as "VOIP call from wifi" they mentioed only this much info. they
> not mentioed info about inbulit SIP client to make voip calls without
> download any third party software.
> as per my search i found this 3 phones E51,E61i,E65

I have an E90 communicator and it's working just fine.

However, when I first got it, it would crash on an incoming call, however 
it stopped crashing after a few rboots and has been OK since then.

A friend also has an E90, and it hasn't stopped crashing on incoming 
calls.

I think they still have a software rev. or 2 to go before it's going to be 
solid...

Gordon


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Re: [asterisk-users] AGI / Voicemail Que

[Doug Lytle]

Nitesh Divecha wrote:
> ([EMAIL PROTECTED]|b)
>
> Any suggestions... By the way I am running Asterisk 1.2.18
>
>   

I believe under 1.2.x it would be [EMAIL PROTECTED]

One of my older dial plans lists:

s-BUSY,1,Voicemail([EMAIL PROTECTED])

Doug


-- 
 
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Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[Guillermo Salas M.]

On Sat, 2008-02-23 at 00:03 +0530, amit salunkhe wrote:
> 
> i want to Buy Nokia E series Phone which have InBulit SIP-VOIP
> Calling client so i can make VOIP calls thru that phone. Aslo that
> Phone easly able to register with Asterisk Pbx to recive inter-office
> calls.  i try to search from web & also from Nokia site but they only
> mention this features as "VOIP call from wifi" they mentioed only this
> much info. they not mentioed info about inbulit SIP client to make
> voip calls without download any third party software.  as per my
> search i found this 3 phones E51,E61i,E65


I've one nokia E65 that works very well with my asterisk box.


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563
USA  : 1 360 968 1701

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 5:49 PM, Vieri <[EMAIL PROTECTED]> wrote:


>  Thanks. I'll try that although I hope it won't go into
>  an infinite loop between the 2 servers.

You are right. That could happen if the phone is not registered anywhere


You can put some security in the dialplan.
 if calls comes from IAX it means that PHONE is not registered in the
other server.
Just create special extensions to take the IAX calls (instead of GoTo):

PHONE  is 101

SERVER 1

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER2/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup

SERVER 2

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER1/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup


I hope it helps,


-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[Michael Iedema]

On 2/22/08, amit salunkhe <[EMAIL PROTECTED]> wrote:
> Hi
> i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
> client so i can make VOIP calls thru that phone. Aslo that Phone easly able
> to register with Asterisk Pbx to recive inter-office calls.
> i try to search from web & also from Nokia site but they only mention this
> features as "VOIP call from wifi" they mentioed only this much info. they
> not mentioed info about inbulit SIP client to make voip calls without
> download any third party software.
>  as per my search i found this 3 phones E51,E61i,E65
>

I have an E65 and it works quite well with my Asterisk install. No
extra software needed, everything is built into those phones. The
menus aren't exactly intuitive to get it setup the first time and
roaming between access points could be better. All in all I'm happy
though.

If you do get an E65, make sure you update your firmware to
2.0633.65.01 or greater. It solves a large number of SIP related
problems.

-Michael

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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 5:49 PM, Vieri <[EMAIL PROTECTED]> wrote:
>
>  --- Andres Jimenez <[EMAIL PROTECTED]> wrote:
>
>  > On Fri, Feb 22, 2008 at 11:42 AM, Vieri
>  > <[EMAIL PROTECTED]> wrote:
>  >
>  > >  However, say ext. 4001 is registered on *1 and
>  > 4002 is
>  > >  registered on *2, if 4001 tries to call 4002 then
>  > I
>  > >  would like to do something like:
>  > >  - lookup 4002 on *1, try to establish a call if
>  > it's
>  > >  REGISTERED here
>  > >  - if it's not registered here then try to look it
>  > up
>  > >  on *2 and establish the call there
>  >
>  > You can do that using the dial plan.
>  >
>  > - Create an IAX link between both servers
>  > - DIal plan in both servers:
>  > First priority Dial using SIP/EXTEN
>  > Second priority IAX/EXTEN Dial IAX/EXTEN
>
>  Thanks. I'll try that although I hope it won't go into
>  an infinite loop between the 2 servers.
>
>
>
>
>   
> 
>  Looking for last minute shopping deals?
>  Find them fast with Yahoo! Search.  
> http://tools.search.yahoo.com/newsearch/category.php?category=shopping
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-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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[asterisk-users] AGI / Voicemail Que

[Nitesh Divecha]

Hello All,

I have my own AGI script running and I am trying to push the call to 
voice mail when Busy, Unavailable and Not Answered.

Everything is working fine but the only problem is voice mail greetings 
for Busy and Unavailable is not played. By default only "Temp Greetings" 
voice mail greetings is played. I am passing the correct parameters for 
Busy => 'b', Unavailable => 'u' and default goes to "Not Answered".

Here is a code sample: -

exec("VOICEMAIL",  $arr);
} elseif ($status == "CHANUNAVAIL"){
$arr = array("[EMAIL PROTECTED]", 'u');
$agi->exec("VOICEMAIL",  $arr);
} else {
$arr = array("[EMAIL PROTECTED]");
$agi->exec("VOICEMAIL",  $arr);
}
?>

Here is the AGI Debug message: -
-- AGI Script Executing Application: (VOICEMAIL) Options: 
([EMAIL PROTECTED]|b)
-- Playing '/var/spool/asterisk/voicemail/default/2481237766/temp' 
(language 'en')

As you can see I am passing the correct parameter for BUSY => |b, but 
Asterisk is only playing the "temporary greetings".

Any suggestions... By the way I am running Asterisk 1.2.18

Cheers,
Nitesh



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[asterisk-users] NOKIA E series Phone for SIP-VOIP calling

[amit salunkhe]

Hi
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
client so i can make VOIP calls thru that phone. Aslo that Phone easly able
to register with Asterisk Pbx to recive inter-office calls.
i try to search from web & also from Nokia site but they only mention this
features as "VOIP call from wifi" they mentioed only this much info. they
not mentioed info about inbulit SIP client to make voip calls without
download any third party software.
 as per my search i found this 3 phones E51,E61i,E65

So Guys plz help me for this.

Regards
Amit
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[asterisk-users] FXO Cards - T38

[Fernando Berretta]

Hi,

Could some one let me know if a fax is received through a FXO card 
connected to * and fax machine is connected to a Linksys FXS device 
which support T38, is T38 going to be used for faxes which comes from 
PSTN or go through PSTN ? or because of Asterisk T38 passthrough support 
it is not possible ? so is for this scenery better to use external FXO 
gateways with t38 support ?

Regards,
Fernando

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Re: [asterisk-users] problem transferring calls some of the times

[Mojo with Horan & Company, LLC]

Sorry, I jut got your other message stating the steps your boss' 
secretary uses to transfer calls, so this question's time is past.

I'm curious if the 'flash' button is the only way those phones can do a 
transfer.  Do they have any other transfer keys, or could you try the 
featuremap codes?  Our polycom transfer buttons have always just worked, 
but my users, for some reason, all felt more comfortable using DTMF 
keypresses...  dunno why :)

So we all press ## to do a blind transfer now, or ** to auto-park to 
first parking space.

Moj

Mojo with Horan & Company, LLC wrote:
> Are you using buttons on your phone to effect the transfer, or are you 
> using codes defined in features.conf?
>
> Moj
> Ian wrote:
>   
>> Hi,
>>
>> Mojo with Horan & Company, LLC said the following on 20-Feb-08 09:31 PM:
>> 
>>> Is it AFTER you have parked a call?  Meaning, for example, you transfer 
>>> an incoming call to 700.  No problem.  Later, when it's picked up from 
>>> 701, can it NOT be transferred again? 
>>>
>>> Moj
>>>   
>>>   
>> No I don't park the call.
>>
>> The call comes in, and gets redirected to our receptionists phone, 
>> from there it gets transferred to another extension (the bosses 
>> secratary) and then gets transferred (to the boss). now the problem, 
>> sometimes that transfer fails, other times the call dont even want to 
>> leave the receptionists phone.
>>
>> The big thing about this problem is that it comes and goes, like 
>> yesterday we didn't have a problem, and I did not change a thing.
>>
>> Ian
>> 
>>> Ian wrote:
>>>   
>>>   
 Hi All

 Sorry to be a bother again but seems like I just cant get away from 
 the problems.

 This time my problem is that *sometimes* a user cant transfer a call 
 from one extension to another, I have narrowed down the problem to it 
 only happening to calls from outside the internal system.

 The wierd thing about the problem is that it comes and goes one moment 
 the user can transfer, and the next call he can't.

 I am running:

 * Asterisk 1.4.17
 * Zaptel 1.4.7.1
 * Libpri 1.4.3

 Using the following phones and firmware

 * Grandstream GXP2000 (with ext pad) : 1.1.4.14
 * Grandstream BT200 : 1.1.4.18

 I have set up the phones to log debug logs to a syslog server, I am 
 still trying to figure out what exactly the log says.

 Is it an * problem, or Grandstream problem

 Does anyone know if I am able to see the keysequence the user types 
 into the phone (just in case it might even be a user made problem), I 
 have tried scanning though the logs of a failed call, but could not 
 see any lines that can be a keypress, or maybe I am looking in the 
 incorrect spot?

 Your help will be greatly appreciated.

 Let me know if, in any way, I can shed some more light on the subject.

 Thanks in advance
 Ian
 -- 
 www.vddi.co.za 
 I Coetzee
 IT Tegnikus
 Telefoon   :   012 664 2300
 Selfoon:   079 522 6519
 Faks   :   012 644 2902
 E-pos  :   [EMAIL PROTECTED]
 Skype  :   vddb_igcoetzee

 

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>>>
>>>
>>>   
>>>   
>> 
>>
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>
>
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Re: [asterisk-users] load balancing SIP extensions

[Vieri]

--- Andres Jimenez <[EMAIL PROTECTED]> wrote:

> On Fri, Feb 22, 2008 at 11:42 AM, Vieri
> <[EMAIL PROTECTED]> wrote:
> 
> >  However, say ext. 4001 is registered on *1 and
> 4002 is
> >  registered on *2, if 4001 tries to call 4002 then
> I
> >  would like to do something like:
> >  - lookup 4002 on *1, try to establish a call if
> it's
> >  REGISTERED here
> >  - if it's not registered here then try to look it
> up
> >  on *2 and establish the call there
> 
> You can do that using the dial plan.
> 
> - Create an IAX link between both servers
> - DIal plan in both servers:
> First priority Dial using SIP/EXTEN
> Second priority IAX/EXTEN Dial IAX/EXTEN

Thanks. I'll try that although I hope it won't go into
an infinite loop between the 2 servers.



  

Looking for last minute shopping deals?  
Find them fast with Yahoo! Search.  
http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [asterisk-users] load balancing SIP extensions

[Yehavi Bourvine +972-8-9489444]

> What I would like to do is have two identical *
> servers which accept registrations of sip extensions
> 4000-4999.
>
> If I define a rrDNS or LinuxHA then I should have
> load-balanced registrations.
>
> However, say ext. 4001 is registered on *1 and 4002 is
> registered on *2, if 4001 tries to call 4002 then I
> would like to do something like:
> - lookup 4002 on *1, try to establish a call if it's
> REGISTERED here
> - if it's not registered here then try to look it up
> on *2 and establish the call there
>...

I've tried doing something similar and came with two options. The common to
them is that I use MySQL for realtime extensions, and set "systemname"
parameter to the IP address of the server where the phone registers.

When a call arrives I check whether the REGSERVER coloumn is the same as the
local server or not. If not, then there are two options:

- Pass the call via IAX to the other servers; this makes both server process
  the call and the audio.

- Send a "refer" message to the caller to contact the other server.

I had this working in the lab but not in production yet. If you want the
dialplan code for this then email me.

   __Yehavi:

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Re: [asterisk-users] Linksys SPA-942 Phones

[C F]

On Fri, Feb 22, 2008 at 7:56 AM, Anciso, Roy <[EMAIL PROTECTED]> wrote:
>
>
>
> Hello List,
>
> After seeing a few positive responses for the Linksys SPA-942 phones I was
> hoping to get some answers on the following questions:
>
> ·   How do the phones handling system wide paging? Is it similar to the
> Polycom phones?

Yes it's quite similar, just be aware that unlike the Polycoms, if you
page a phone (auto answer header) that is in a conversation the SPAs
will put the current call on hold and take the auto answered phone,
therefore before paging an SPA make sure to use app_chanisavail first.


> ·   Can a corporate directory be configured with the phones using
> Asterisk?

Don't know, I never tried it, but IIRC the web page to the SPA configs
has a tab for that.

> ·   How is the speaker phone quality?

For the price it's great, but it's not the best, in my opinion it's
acceptable quality. It's not as good as the Cisco which is worst than
the Polycoms. I would say it's around as good as the Aastras


>
> Thanks
>
> Roy Anciso
>
> Director of Technology
>
> Manistee Intermediate School District
>
> 772 East Parkdale Avenue
>
> Manistee, MI 49660
>
> Ph: 231-723-4264
>
> Fx: 231-398-3036
>
> [EMAIL PROTECTED]
>
>
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Re: [asterisk-users] problem transferring calls some of the times

[Mojo with Horan & Company, LLC]

Are you using buttons on your phone to effect the transfer, or are you 
using codes defined in features.conf?

Moj
Ian wrote:
> Hi,
>
> Mojo with Horan & Company, LLC said the following on 20-Feb-08 09:31 PM:
>> Is it AFTER you have parked a call?  Meaning, for example, you transfer 
>> an incoming call to 700.  No problem.  Later, when it's picked up from 
>> 701, can it NOT be transferred again? 
>>
>> Moj
>>   
> No I don't park the call.
>
> The call comes in, and gets redirected to our receptionists phone, 
> from there it gets transferred to another extension (the bosses 
> secratary) and then gets transferred (to the boss). now the problem, 
> sometimes that transfer fails, other times the call dont even want to 
> leave the receptionists phone.
>
> The big thing about this problem is that it comes and goes, like 
> yesterday we didn't have a problem, and I did not change a thing.
>
> Ian
>> Ian wrote:
>>   
>>> Hi All
>>>
>>> Sorry to be a bother again but seems like I just cant get away from 
>>> the problems.
>>>
>>> This time my problem is that *sometimes* a user cant transfer a call 
>>> from one extension to another, I have narrowed down the problem to it 
>>> only happening to calls from outside the internal system.
>>>
>>> The wierd thing about the problem is that it comes and goes one moment 
>>> the user can transfer, and the next call he can't.
>>>
>>> I am running:
>>>
>>> * Asterisk 1.4.17
>>> * Zaptel 1.4.7.1
>>> * Libpri 1.4.3
>>>
>>> Using the following phones and firmware
>>>
>>> * Grandstream GXP2000 (with ext pad) : 1.1.4.14
>>> * Grandstream BT200 : 1.1.4.18
>>>
>>> I have set up the phones to log debug logs to a syslog server, I am 
>>> still trying to figure out what exactly the log says.
>>>
>>> Is it an * problem, or Grandstream problem
>>>
>>> Does anyone know if I am able to see the keysequence the user types 
>>> into the phone (just in case it might even be a user made problem), I 
>>> have tried scanning though the logs of a failed call, but could not 
>>> see any lines that can be a keypress, or maybe I am looking in the 
>>> incorrect spot?
>>>
>>> Your help will be greatly appreciated.
>>>
>>> Let me know if, in any way, I can shed some more light on the subject.
>>>
>>> Thanks in advance
>>> Ian
>>> -- 
>>> www.vddi.co.za 
>>> I Coetzee
>>> IT Tegnikus
>>> Telefoon:   012 664 2300
>>> Selfoon :   079 522 6519
>>> Faks:   012 644 2902
>>> E-pos   :   [EMAIL PROTECTED]
>>> Skype   :   vddb_igcoetzee
>>>
>>> 
>>>
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>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>> 
>>
>>
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>>
>>
>>   
>
> 
>
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Re: [asterisk-users] (no subject)

[C F]

vi /etc/asterisk/extensions.conf

On Fri, Feb 22, 2008 at 12:08 AM, sandeep <[EMAIL PROTECTED]> wrote:
>
>
>
> hi,
>
> how to write a advanced dial plan
>
> for example:
> dial to a extension(123).if the user didnot pick the call, caller should get
> a ivr script(Enter 1 to to dial operator  and 2 to go to voicemail)
> If caller press 1 it should dial to the operator,else if he dials 2 it
> should go to the voicemail of calle's extension.
>
> thanks
> sandeep.
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Re: [asterisk-users] Voted most stable and easy to use phone?

[shadowym]

I guess someone has to say it.
 
Have you considered Aastra?
 
You can argue about quality/features/functionality but I have set up both
and the Aastra are definitely easier to configure and they reboot quicker.  
Nobody ever complains about the quality of sound or speakerphone on them
either.
 
 
From: John Faubion [mailto:[EMAIL PROTECTED] 
Sent: Thursday, February 21, 2008 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voted most stable and easy to use phone?
 
>A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing  
>issues and we all discussed the problems we had with the 360s and some
suggestions were  made but the   
>new polycom phones had just hit the market and not many people were able to
comment on them. 
 
We just installed a dozen of the Polycom IP-330 phones. Initially out of the
box I wasn't real sure about the decision to use them. The phones are very
small and don't seem to have very many features. However in use they have
been great. They don't waste a lot of desk space, they don't overwhelm the
users and they seem to provide adequate information. They're easy to use and
Polycom reliable. The speaker phone is still really good though I'm not sure
it is as good as the 501/601 phones. I haven't really done a side by side
comparison of that but I think the 501/601 has a better speaker phone. I
can't see buying another GXP after using these. The difference in price just
isn't worth the aggravation.
 
John
 
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Re: [asterisk-users] asterisk config file online editor

[Anton Krall]

With some mods it surely did the trick

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan 
& Company, LLC
Sent: miércoles, 20 de febrero de 2008 01:49 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor

No problem, hope it gets you where you need to be :)

Moj

Anton Krall wrote:
> This is a good start, thx Moj
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
> Horan & Company, LLC
> Sent: martes, 19 de febrero de 2008 01:35 p.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] asterisk config file online editor
>
> Like 15 lines of php and html?
>
>  $fn = "/etc/asterisk/extensions.conf";
>
> if ($_REQUEST['action'] == "write" && $_REQUEST['contents'] != "")
> {
> rename($fn, $fn.".".date("U"));
> $fp = fopen($fn, "wt");
> fwrite($fp, $_REQUEST['contents']);
> fclose($fp);
> }
>
> ?>
> 
> 
> 
> 
>  
> 
>
> Security holes galore!  clean it up a bit :)  And check on permissions 
> issues, that your httpd can write to the file.
>
> Moj
>
> Anton Krall wrote:
>   
>> Guys, Im looking for a good text file editor for asterisk config files
>> that can be embedded on a web page for online editing (on an
>> 
> interface),
>   
>> any recommendations?
>>
>>
>> 
>> Anton Krall
>> Direccion General
>>
>> Intruder Consulting
>> A Division of IntruderEnterprises S.A. de C.V.
>> www.Intruder.com.mx
>> www.IntruderStore.com.mx
>>  
>> Tel. 3872-2200 ext. 201
>> Tel. 01-800-INTRUDER (01-800-468-7833)
>> Email: [EMAIL PROTECTED]
>>
>> Como lo estoy haciendo? Contacte a mi Director:
>>
>>
>>
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>
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Re: [asterisk-users] High CPU load after upgrading to 1.4

[shadowym]

Did you file a bug report?
http://bugs.digium.com


-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED] 
Sent: Thursday, February 21, 2008 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4

On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote:
> I currently have 1558 sip peers loaded in
> Asterisk and the current CPU load is 10% when no calls are being
> processed and no sip registrations.

At first glance, I would think that maybe you have "qualify=yes" in each
of your SIP peers, which is keeping Asterisk busy checking to see if the
peers are responding or not.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.






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Re: [asterisk-users] Interrupt VM and Steal a call.

[Robert Lister]

On Fri, Feb 22, 2008 at 09:05:17AM -0500, Michael Munger wrote:
> Two questions:
> 
> 1.   Does anyone have a good way to transfer a call from inside
> comedian mail to the current extension? The problem is: let's say the
> phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd
> ring. I come running into my office but miss it by a split second. Is
> there a way I can barge in on the person leaving a message for my
> mailbox while they're leaving it?

I imagine that would be tricky to do once the call has been handed in to the 
voicemail application, as presently you are limited in what you can do to 
the call once it's gone in there. You might be able to locate the original 
SIP channel and bridge the calls, but I've no idea how you would track that 
properly. There is a way to make voicemail have a "press 0 to be transferred 
to somewhere else" option. We use that here and it works. Users can set up 
where they want the caller to be transferred to (usually a mobile) and then 
they can record on their outgoing message "leave me a message, or press 0 to 
try my mobile..."

Or, Sounds like a case for a few IP DECT cordless handsets to save all this 
running about! You might run into somebody carrying a boiling hot cup of 
coffee in your rush to answer the phone! (happens!)

We have a few Siemens C460 IP DECT Phones. The range and battery life on 
them is by far superior to any of the WiFi/SIP phones I've tried so far. I 
have a SIP/Wifi Nokia E65 that works great, but the battery life is not very 
good when the wifi is left on, and it was less than straightforward to set 
up!

> 2.   If a phone rings a receptionist desk, and the receptionist is
> down the hall, she wants to be able to dial an extension, and have that
> transfer the call from her desk to the phone she's currently on so she
> doesn't have to run to her desk. Is there a built in feature for this or
> do I have to code it?

There is a feature called pickup defined in features.conf:

pickupexten = *8

Restart asterisk if you need to change features.conf (in my experience just 
a reload when changing features.conf doesn't always work)

You then need to define your SIP/devices into pickup groups in sip.conf, for 
example:-

[500]
canreinvite=yes
nat=no
secret=...
dtmfmode=rfc2833
callgroup=1
pickupgroup=1

[501]
canreinvite=yes
nat=no
secret=...
dtmfmode=rfc2833
callgroup=1
pickupgroup=1

Then reload. Now, if extn 500 were ringing, picking up 501 and doing *8 will 
connect that call. 

For busier systems I believe there is a dialplan feature that enables 
"directed" pickup so you can pick up a specific extension, but I haven't 
played with that so I can't say how it works. That might be more suitable.

The callgroup defines what pickup group the device is in, and pickupgroup 
defines what groups (when that extension dials *8) that device can pick up.
A device can be in one callgroup but multiple pickup groups:-

pickupgroup=1,2

This is so that if you have many sites or departments, only people who sit 
within the range of the ringing phone can pick it up, and not get connected 
to some other random call incoming somewhere else.


Rob


-- 
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510

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Re: [asterisk-users] Interrupt VM and Steal a call.

[Doug Lytle]

Michael Munger wrote:
>
> Two questions:
>
> 1. Does anyone have a good way to transfer a call from inside comedian 
> mail to the current extension? The problem is: let’s say the phone 
> goes to voicemail after 4 rings, yet I don’t hear it until the 3^rd 
> ring. I come running into my office but miss it by a split second. Is 
> there a way I can barge in on the person leaving a message for my 
> mailbox while they’re leaving it?
>
This can be done with FOP (Flash Operator Panel) or any application that 
can issue Manager commands to Asterisk.


> 2. If a phone rings a receptionist desk, and the receptionist is down 
> the hall, she wants to be able to dial an extension, and have that 
> transfer the call from her desk to the phone she’s currently on so she 
> doesn’t have to run to her desk. Is there a built in feature for this 
> or do I have to code it?
>

Yes, this option is accessed via the features.conf file. It's the 
pickupexten feature.

Doug


-- 
 
Ben Franklin quote:

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



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Re: [asterisk-users] How to get a clean, basic configuration?

[Tilghman Lesher]

On Friday 22 February 2008 04:55:13 Vincent wrote:
> On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote:
> >For the brave: use modules.conf without 'autoload = yes'. This promises
> >you many hours of interesting dialplan debugging. Enjoy.
>
> Yup, that's what I anticipated, which is why I was asking which
> modules I can _safely_ remove without breaking things :-)

Generally, the rule is that you can't remove any of the res_*
modules.  If you follow that rule, you can then start to work out the
applications, functions, and channels that you don't want.  The codecs
and formats modules are usually too small to consider disabling, unless
it is absolutely critical for space (such as on embedded systems, where
you generally don't want any transcoding anyway).

-- 
Tilghman

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Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[Tzafrir Cohen]

On Fri, Feb 22, 2008 at 07:59:14AM -0500, Matt Florell wrote:
> Hello,
> 
> This was about a year ago when we abandoned putting new systems on the
> 2.4.33 kernel. About that time we started having other vendors stop
> supporting it very well also so it wasn't only that card that we were
> having issues with. The problems were related to the CRC Linux modules
> and zaptel not liking them in 2.4.

Try again, then. In the recent monthes there were some 2.4-specific
compilation fixes.

> 
> One very interesting side note, on identical hardware we got 5-10%
> better Asterisk performance on a 2.4 kernel than we got on 2.6
> kernels.

Interesting regression. Zaptel is generally a program that stresses some
parts of the ernel in unexpected ways.

> 
> Overall we are happy with 2.6 and it's new features and we don't
> really want to go back at this point. I just wanted to mention our
> experiences with kernel problems.

Good and detailed bug reports are always welcomed.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Can asterisk support 20 user's conference?

[Tzafrir Cohen]

On Fri, Feb 22, 2008 at 03:22:39PM +0100, Patrick wrote:

> > And I want to build HD audio conference by using polycom’s 650 ip phone. 
> > Can asterisk support G722 HD audio conference?
> 
> Afaik Asterisk only supports it in 1.6beta. If you need a working
> solution *now* then have a look at FreeSWITCH which supports wideband
> and ultra-wideband conferences very well.

Both Asterisk 1.6 and FreeSwitch are not officially a stable release.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Digium B410P and 8 ports connectivity

[Andres]

Olivier wrote:

> Hello,
>
> Junghanns and BeroNet offer 8 BRI ports cards.
>
> Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI 
> ports subsystem (2 PCI slots used, but 1 one set of interrupts) could 
> be made out of 2 B410P ?

You can also use the Sangoma A500 for up to 24 BRI ports.  You get 6 
ports on the main card and 6 on every remora.  Only 1 PCI slot is used.  
The remoras don't require the slot, just the physical space.

Andres.
http://www.neuroredes.com

>
> I know you could (theorically) do this with Junghanns and BeroNet cards.
>
> Cheers
>
>
>
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Re: [asterisk-users] which codec over iax => pstn

[sean darcy]

Atis Lezdins wrote:

>>>
 BTW, we have 512kbs over the iax connection.
>>> G711 needs about 80Kb/sec each way to work. (It's 64Kb/sec plus IP
>>> overhead). GSM needs about 32Kb/sec (13Kb/sec plus IP overhead).
>>>
>> So with DSL 512kbs up and 3mbs down, plenty of room for G711.
> 
> Take the weakest link - up 512 kbps, that makes 6 simultenous ulaw
> calls (not counting other traffic). Of course you could push more
> calls, inbound voice would be good (3mbps) but outbound would be
> crappy.
> 
>> Of course, I could figure out how to configure QOS in iptables for
>> asterisk, it'd be a lot better.
> 
> If you have fixed bandwidth, it should be fairly simple, there's some
> ready scripts for scheduling outbound/inbound traffic on fixed
> bandwidth links. This is a very good resource for that -
> http://lartc.org/
> 
> Also i found this yesterday, could be good for start. It doesn't
> assume fixed bandwidth, but just gives priority to VoIP packets.
> http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk
> 
That's a great find. I've never been able to figure out the lartc docs. 
This really lays it out, simply. Should be linked from the digium or 
voip-info sites.

> Regards,
> Atis
> 


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Re: [asterisk-users] Can asterisk support 20 user's conference?

[Patrick]

On Thu, 2008-02-21 at 13:57 +0800, zhao_x_q wrote:
> HI, Friends,
> 
>  Now I have 20 polycom’s SS2 phones. Can Asterisk support 20
> users conference meeting? 

Yes.

> And I want to build HD audio conference by using polycom’s 650 ip phone. 
> Can asterisk support G722 HD audio conference?

Afaik Asterisk only supports it in 1.6beta. If you need a working
solution *now* then have a look at FreeSWITCH which supports wideband
and ultra-wideband conferences very well.

Regards,
Patrick



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Re: [asterisk-users] (no subject)

[Jared Smith]

On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
> for example:
> dial to a extension(123).if the user didnot pick the call, caller
> should get a ivr script(Enter 1 to to dial operator  and 2 to go to
> voicemail)
> If caller press 1 it should dial to the operator,else if he dials 2 it
> should go to the voicemail of calle's extension.

It's really pretty easy.  

; Call the SIP peer, let the phone ring for 20 seconds
exten => 123,1,Dial(SIP/some_sip_peer,20)
; Play the "press-1-or-press-2" prompt, get one digit
; from the caller, and save it to a variable called
; ${option}
exten => 123,n,Read(option,press-1-or-press-2,1)
; If the caller enters 1, send the call to the [some_context] context,
; to the "operator" extension, priority 1
exten => 123,n,GotoIf($["${option}" = "1"]?some_context,operator,1)
; Otherwise, send the  call to voicemail
exten => 123,n,VoiceMail([EMAIL PROTECTED])

I haven't actually taken the time to test this in my own dialplan, but
it should work.  Obviously you'll want to change the name of the SIP
peer you're dialing, as well as the location of the operator extension.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Interrupt VM and Steal a call.

[Michael Munger]

Two questions:

 

1.   Does anyone have a good way to transfer a call from inside
comedian mail to the current extension? The problem is: let's say the
phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd
ring. I come running into my office but miss it by a split second. Is
there a way I can barge in on the person leaving a message for my
mailbox while they're leaving it?

2.   If a phone rings a receptionist desk, and the receptionist is
down the hall, she wants to be able to dial an extension, and have that
transfer the call from her desk to the phone she's currently on so she
doesn't have to run to her desk. Is there a built in feature for this or
do I have to code it?

 

Yours,

Michael Munger, dCAP

404-438-2128

[EMAIL PROTECTED]  

 

Attachment encrypted? click here
 .

 

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Re: [asterisk-users] Will this be sufficient for 20+ concurrent calls?

[Jared Smith]

On Fri, 2008-02-22 at 14:51 +0100, harry wrote:
> This is my first time setting up Asterisk in production and we are
> buying the Digium TE121-card for use with an ISDN-30 connection. We
> are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
> http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html
> - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333
> Mhz. 2 GB RAM, 3.5" SATA II discs. I know this is a rather vague
> question that can not be precisely answered, but mainly handling calls
> w/ playback of local gsm-files - that is, no redirection to other
> clients etc. - will it be sufficient for 20-something concurrent
> calls?

That server should be more than adequate for 20 concurrent calls.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Will this be sufficient for 20+ concurrent calls?

[Raúl Gómez C.]

Harry,

I think this system will suffice for your needs. I have a similar setup
working great with 2 Dual Core Xeon @ 2GHz

On Sat, Feb 23, 2008 at 9:21 AM, harry <[EMAIL PROTECTED]> wrote:

> This is my first time setting up Asterisk in production and we are
> buying the Digium TE121-card for use with an ISDN-30 connection. We
> are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
>
> http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html
> - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333
> Mhz. 2 GB RAM, 3.5" SATA II discs. I know this is a rather vague
> question that can not be precisely answered, but mainly handling calls
> w/ playback of local gsm-files - that is, no redirection to other
> clients etc. - will it be sufficient for 20-something concurrent
> calls?
>
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-- 
Nacho
Linux Counter #156439
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Re: [asterisk-users] [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?

[Jared Smith]

On Fri, 2008-02-22 at 07:43 -0600, Michael Graves wrote:
> Let me add another variable into the mix...what about the Linksys
> SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a
> deal at $80 street price.

I'm quite happy with my SPA-962 + sidecar... I tend to use it more than
the other phones I have here in my hardware zoo.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] High CPU load after upgrading to 1.4

[xrem1x]

I verified that qualify=no.  I am getting this CPU load at 10% without any 
peers even registered which is very strange, but it doesn't happen when I run 
on 64-bit CentOS 5 kernel. Remi

- Original Message -
From: Jared Smith 
Date: Thursday, February 21, 2008 5:55 pm
Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4
To: Asterisk Users Mailing List - Non-Commercial Discussion 

> On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote:
> > I currently have 1558 sip peers loaded in
> > Asterisk and the current CPU load is 10% when no calls are being
> > processed and no sip registrations.
> 
> At first glance, I would think that maybe you have "qualify=yes" 
> in each
> of your SIP peers, which is keeping Asterisk busy checking to 
> see if the
> peers are responding or not.
> 
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
> 
> 
> 
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[asterisk-users] adjusting volume on a wildcard 100XP with zaptel's {t, r}xgain

[Brian J. Murrell]

I'm finding the volume of the calls on my wildcard 100XP (clone) is too
low.  I understand I can muck with rxgain and/or txgain (which one in
fact will increase the volume of the other party as far as I'm hearing
it?) to deal with this but right now I have them both at 0.00 and I am
concerned about introducing echo by fiddling with them given that these
two registers are what are fiddled with to deal with echo problems too.

How can I safely adjust these?  Caller report low volume problems from
me too, so if I make equal adjustments to both will that maintain my
level of (non-)echo?

Thanx,
b.



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Re: [asterisk-users] Will this be sufficient for 20+ concurrent calls?

[Julian Lyndon-Smith]

More than enough.

Julian.

harry wrote:
> This is my first time setting up Asterisk in production and we are
> buying the Digium TE121-card for use with an ISDN-30 connection. We
> are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
> http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html
> - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333
> Mhz. 2 GB RAM, 3.5" SATA II discs. I know this is a rather vague
> question that can not be precisely answered, but mainly handling calls
> w/ playback of local gsm-files - that is, no redirection to other
> clients etc. - will it be sufficient for 20-something concurrent
> calls?
> 
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[asterisk-users] Will this be sufficient for 20+ concurrent calls?

[harry]

This is my first time setting up Asterisk in production and we are
buying the Digium TE121-card for use with an ISDN-30 connection. We
are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html
- for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333
Mhz. 2 GB RAM, 3.5" SATA II discs. I know this is a rather vague
question that can not be precisely answered, but mainly handling calls
w/ playback of local gsm-files - that is, no redirection to other
clients etc. - will it be sufficient for 20-something concurrent
calls?

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Re: [asterisk-users] [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?

[Michael Graves]

Let me add another variable into the mix...what about the Linksys
SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a
deal at $80 street price.

Michael

On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote:

>
>Michael Graves wrote:
>> I need to add a few phones to an existing installation. They have a
>> dozen IP430 at the moment. Does anyone feel that there are advantages
>> to the IP330? Cost is not the major consideration as long as they're in
>> the same range. (under $175)
>> 
>> Michael
>> --
>> Michael Graves
>> mgravesmstvp.com
>> blog.mgraves.org
>> o713-861-4005
>> c713-201-1262
>> sip:[EMAIL PROTECTED]
>> skype mjgraves
>> fwd 54245
>> 
>> 
>> 
>> > 
>
>
>The 430 has a larger footprint and feels to be a slightly heftier and
>more business-like phone.  I have very similar sound quality and results
>with either phone.  To me it would be a toss-up based upon the
>customer's taste.
>
>-- 
>Join the FSF as an Associate Member at:
>http://www.fsf.org/register_form?referrer=5175>
>
>
>
>
>--~--~-~--~~~---~--~~
>Your participation in the conference is always appreciated! Please try to be 
>there live when it happens.
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>

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245



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[asterisk-users] Opinions please: Polycom IP 430 vs 330?

[Michael Graves]

I need to add a few phones to an existing installation. They have a
dozen IP430 at the moment. Does anyone feel that there are advantages
to the IP330? Cost is not the major consideration as long as they're in
the same range. (under $175)

Michael
--
Michael Graves
mgravesmstvp.com
blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245



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[asterisk-users] DUNDi ${NUMBER} variable not defined

[Vieri]

If one does a "dundi lookup", shouldn't the ${NUMBER}
variable be replaced with the current value?
ie. if I run "dundi lookup [EMAIL PROTECTED]" shouldn't I get
an answer string like
"IAX2/priv:[EMAIL PROTECTED]/4065 (EXISTS)"?

The *CLI does not show me the dst extension:

*CLI> dundi lookup [EMAIL PROTECTED]
  1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
 from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 53 ms

*CLI> dundi lookup [EMAIL PROTECTED]
  1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
 from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 37 ms

What could be wrong?




  

Never miss a thing.  Make Yahoo your home page. 
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Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[Matt Florell]

Hello,

This was about a year ago when we abandoned putting new systems on the
2.4.33 kernel. About that time we started having other vendors stop
supporting it very well also so it wasn't only that card that we were
having issues with. The problems were related to the CRC Linux modules
and zaptel not liking them in 2.4.

One very interesting side note, on identical hardware we got 5-10%
better Asterisk performance on a 2.4 kernel than we got on 2.6
kernels.

Overall we are happy with 2.6 and it's new features and we don't
really want to go back at this point. I just wanted to mention our
experiences with kernel problems.

Thanks,

MATT---



On 2/22/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Fri, Feb 22, 2008 at 02:33:01AM -0500, Matt Florell wrote:
>  > Hello,
>  >
>  > I was never able to get the TE407P card running on a 2.4 Linux kernel.
>  > Using a 2.6 kernel I was able to get it working.
>
>
> What error(s) do you get? On what platform / kernel exactly?
>
>
>  >
>  > Not really surprising since a lot of companies do not support or even
>  > test on Linux 2.4 any more.
>
>
> Well, nobody really likes 2.4, I guess.
>
>  Build logs of zaptel svn on various kernels from recent days. A certain
>  2.4 is included there:
>
>  http://updates.xorcom.com/logs/zaptel/report.html
>
>
>  --
>Tzafrir Cohen
>  icq#16849755  jabber:[EMAIL PROTECTED]
>  +972-50-7952406   mailto:[EMAIL PROTECTED]
>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
>
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[asterisk-users] Linksys SPA-942 Phones

[Anciso, Roy]

Hello List,
After seeing a few positive responses for the Linksys SPA-942 phones I
was hoping to get some answers on the following questions:
*   How do the phones handling system wide paging? Is it similar to
the Polycom phones?
*   Can a corporate directory be configured with the phones using
Asterisk? 
*   How is the speaker phone quality? 

Thanks

Roy Anciso 
Director of Technology
Manistee Intermediate School District
772 East Parkdale Avenue
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-398-3036
[EMAIL PROTECTED]

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Re: [asterisk-users] Pattern matching....

[Michael Munger]

That's what I figured, and the way I've always done it. I was just
*hoping* someone knew of a better way that I didn't know about.

Yours,

Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: Thursday, February 21, 2008 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pattern matching

On Thu, 2008-02-21 at 13:34 -0500, Mike Trest - Personal wrote:
> [746][704][048]

Nope, that's not going to do exactly what you want either... that
pattern would match a lot of area codes besides the ones you're looking
for.  (For example, you could have 7 as the first digit and 0 as the
second digit and 0 as the third digit.)

You really need to have three separate patters; one for each area code.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 11:42 AM, Vieri <[EMAIL PROTECTED]> wrote:

>  However, say ext. 4001 is registered on *1 and 4002 is
>  registered on *2, if 4001 tries to call 4002 then I
>  would like to do something like:
>  - lookup 4002 on *1, try to establish a call if it's
>  REGISTERED here
>  - if it's not registered here then try to look it up
>  on *2 and establish the call there

You can do that using the dial plan.

- Create an IAX link between both servers
- DIal plan in both servers:
First priority Dial using SIP/EXTEN
Second priority IAX/EXTEN



Dial IAX/EXTEN

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] canreinvite question

[Vincent]

On Fri, 22 Feb 2008 18:50:16 +0800, Ron <[EMAIL PROTECTED]> wrote:
>If i set, canreinvite=yes on all ext, assuming all ip phones have the 
>same codec, if 100 calls 101, or vice versa will rtp still go thru 
>asterisk? and same scenario for 200 to 202 or vice versa.

... and I'd like to add to this question: If the phones have the
option "Enable NAT", I expected them to be able to talk to each other
directly, but they didn't, and I had to set them to "canreinvite=no"
in sip.conf. Why is that?


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Re: [asterisk-users] Voted most stable and easy to use phone?

[Rob Hillis]

They have their ups and downs.  If you live outside the US, localising 
your tones is a pain in the proverbial since you have to define every 
tone by frequency combination and intervals, although I guess you do 
only need to do it once.


One other shortcoming of the 942 is the lack of any usable BLF buttons 
at all.  I'm led to believe that one of the upcoming firmware releases 
will allow you to use the four line buttons as BLF buttons, but since 
you need at least two line buttons in order to be able to transfer a 
call, that means you realistically would only get two BLF keys - not 
many - especially if you're used to the Snoms or Grandstreams (hack, spit)


The 962s are /really/ nice phones, albeit somewhat more expensive.  The 
large colour display is /very/ nice, and button banks for the 962s are 
available and very cheap compared to many other options, even if they do 
look a little cheap.



arkda wrote:
I'm a huge fan of the Linksys SPA-942s for users. They run around 
$125, are pretty straightforward to manage via TFTP, and work really 
well with Asterisk.


On Thu, Feb 21, 2008 at 4:07 AM, Michael J. Liberatore 
<[EMAIL PROTECTED] 
> wrote:


A while back i had asked about possible replacements for snom 360
phones that were breaking and causing issues and we all discussed
the problems we had with the 360s and some suggestions were made
but the  new polycom phones had just hit the market and not many
people were able to comment on them.
 
Basically i am looking to get some new phones and in the process

get rid of the countless number of problems i have had that has
always been caused by phones (snom 360's and gxp-2000's).  I would
like to get the feedback of the list on the phone voted best for
stability, working with *, and ease of use for dumb non tech users.
 
I was thinking of trying one of these new polycom phones that are

about $150, but havent gotten any feedback on them yet.
 
Basically i am interested in any phones but snom's, grandstreams,

and sipura's/linksys.  mainly polycom's i guess.
 
thanks
 
mike


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[asterisk-users] load balancing SIP extensions

[Vieri]

What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999. 

If I define a rrDNS or LinuxHA then I should have
load-balanced registrations. 

However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002 then I
would like to do something like:
- lookup 4002 on *1, try to establish a call if it's
REGISTERED here
- if it's not registered here then try to look it up
on *2 and establish the call there

I tried to use DUNDi on my local servers but I can't
seem to make it work. Most howtos out there explain
the use of DUNDi when the extension ranges do not
overlap.
So in my case where both *1 and *2 have the same local
extension range 4XXX, can I go the DUNDi route or
should I stop bashing my head on that and explore
another solution?

If someone has configured a similar system then I'd
greatly appreciate some tips.
I read a few dundi docs like
http://www.voip-info.org/wiki-DUNDi.

Thanks



  

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Re: [asterisk-users] Pattern matching....

[Tony Mountifield]

In article <[EMAIL PROTECTED]>, Eric Wieling <[EMAIL PROTECTED]> wrote:
> No that will not work.  You would want three exten => lines, one for 
> each area code.

And if you have a lot of common dialplan that you don't want to duplicate
between the three extension patterns, put the common stuff up at a higher
priority and use Goto to get there:

exten => _404NXX,1,Goto(200)
exten => _770NXX,1,Goto(200)
exten => _678NXX,1,Goto(200)

exten => _NXXNXX,200,NoOp(Start of common instructions)
exten => _NXXNXX,n,etc

Cheers
Tony

> Michael Munger wrote:
> > Will this work to match any number from the 770,404, or 678 area codes? 
> > 
> >  
> > 
> > _[404|770|678]NXX
> > 
> >  
> > 
> > If this won't work, is there a pattern that will do this?
> > 
> >  
> > 
> > Yours,
> > 
> > Michael Munger, dCAP
> > 
> > 404-438-2128
> > 
> > [EMAIL PROTECTED]  
> > 
> >  
> > 
> > Attachment encrypted? click here
> >  .
> > 
> >  
> > 
> > 
> > 
> > 
> > 
> > 
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> 
> -- 
> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
> T-1, PRI, Frame Relay, Linux, and network design.  Based near 
> Birmingham, AL.  Now accepting clients worldwide.
> 
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-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] How to get a clean, basic configuration?

[Vincent]

On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>For the brave: use modules.conf without 'autoload = yes'. This promises
>you many hours of interesting dialplan debugging. Enjoy.

Yup, that's what I anticipated, which is why I was asking which
modules I can _safely_ remove without breaking things :-)


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Re: [asterisk-users] How to get a clean, basic configuration?

[Vincent]

On Thu, 21 Feb 2008 08:33:20 -0500, "C F" <[EMAIL PROTECTED]> wrote:
>first off I anwered you to use vi and you complained showing me cat.

There's some misunderstanding. I didn't complain. I just didn't know
if Asterisk only looked for stuff in modules.conf because there was so
little there and so much stuff scrolling by when I type "reload".

Before loading modules explicitely, I need to make sure what each does
precisely, and what the interdependencies are, if any, so that I know
what the consequences are if I decide not to load a mdoule that looks
like it's not needed on my setup.

Thanks.


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[asterisk-users] canreinvite question

[Ron]

Hi All,


if i do this setup:

  |---[ext 100]
  |--[router/nat gw]--|
  |   |---[ext 101]
  |
[asterisk]--[internet]---|
  |
  |   |---[ext 200]
  |--[router/nat gw]--|
  |---[ext 201]


If i set, canreinvite=yes on all ext, assuming all ip phones have the 
same codec, if 100 calls 101, or vice versa will rtp still go thru 
asterisk? and same scenario for 200 to 202 or vice versa.

what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go 
thru asterisk?

thank you

regards,
Ron


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[asterisk-users] Digium B410P and 8 ports connectivity

[Olivier]

Hello,

Junghanns and BeroNet offer 8 BRI ports cards.

Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports
subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out
of 2 B410P ?

I know you could (theorically) do this with Junghanns and BeroNet cards.

Cheers
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Re: [asterisk-users] Voted most stable and easy to use phone?

[Mindaugas Kezys]

Linksys SPA942. Tried most of available phones on the market. 

These phones sits on companies tables for more then a year. 

No problem at all, easy to use, nice(!) to use. I recommend to everybody.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, February 22, 2008 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voted most stable and easy to use phone?

On Thu, Feb 21, 2008 at 7:32 PM, arkda <[EMAIL PROTECTED]> wrote:
> I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
> pretty straightforward to manage via TFTP, and work really well with
> Asterisk.

I agree, we've had zero trouble with these. Easy to install and they just work.

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[asterisk-users] Using app_sms in South Africa

[Louwrens Benadé]

Anyone from South Africa out there that has gotten SMS over Telkom lines
right? 

 

I’ve found the SMSC but I don’t have the foggiest how to go about it…

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Re: [asterisk-users] Weird Zaptel sound after anwser calls

[voip crazy]

I forgot to say that I'm using bristuff-0.4.0 with zaptel 1.4.4, libpri
1.4.1 and asterisk 1.4.9

Thanks.


2008/2/22, voip crazy <[EMAIL PROTECTED]>:
>
> Dear list,
>
> We have an weird problem with our FXO card (TDM01B). When I made a call
> using this channel, all goes well, the called phone rings but when the
> called phone answers the call. In me handset I can hear an weird sound like
> a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is
> like,
>
> language=en
> context=from-zaptel
> switchtype=national
> usecallerid=yes
> callerid=asreceived
> transfer=yes
> callreturn=yes
> rxgain=-3.0
> txgain=-3.0
> immediate=no
> busydetect=yes
> busycount=8
> answeronpolarityswitch=yes
> hanguponpolarityswitch=yes
> ringtimeout=8000
> faxdetect=both
> group=0
> signalling=fxs_ks
> channel => 7
>
> think the problem is not by echo cause I use fxotune, and the problem
> persist. I made lots of TDM02B instalations and never get this kind of
> problem.
>
> Any clue will be welcomed.
>
> Thanks in advance.
>
> Regards
>
> VoipCrazy
>
>
>
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[asterisk-users] Weird Zaptel sound after anwser calls

[voip crazy]

Dear list,

We have an weird problem with our FXO card (TDM01B). When I made a call
using this channel, all goes well, the called phone rings but when the
called phone answers the call. In me handset I can hear an weird sound like
a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is
like,

language=en
context=from-zaptel
switchtype=national
usecallerid=yes
callerid=asreceived
transfer=yes
callreturn=yes
rxgain=-3.0
txgain=-3.0
immediate=no
busydetect=yes
busycount=8
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
ringtimeout=8000
faxdetect=both
group=0
signalling=fxs_ks
channel => 7

think the problem is not by echo cause I use fxotune, and the problem
persist. I made lots of TDM02B instalations and never get this kind of
problem.

Any clue will be welcomed.

Thanks in advance.

Regards

VoipCrazy
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Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[Tzafrir Cohen]

On Fri, Feb 22, 2008 at 02:33:01AM -0500, Matt Florell wrote:
> Hello,
> 
> I was never able to get the TE407P card running on a 2.4 Linux kernel.
> Using a 2.6 kernel I was able to get it working.

What error(s) do you get? On what platform / kernel exactly?

> 
> Not really surprising since a lot of companies do not support or even
> test on Linux 2.4 any more.

Well, nobody really likes 2.4, I guess.

Build logs of zaptel svn on various kernels from recent days. A certain
2.4 is included there:

http://updates.xorcom.com/logs/zaptel/report.html

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] How to get a clean, basic configuration?

[Tzafrir Cohen]

On Fri, Feb 22, 2008 at 01:28:58AM -0800, Steve Langstaff wrote:
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Tzafrir Cohen
> 
> > For the brave: use modules.conf without 'autoload = yes'. 
> > This promises you many hours of interesting dialplan debugging. Enjoy.
> 
> Is there any method of automatically parsing a dialplan and generating a
> list of the modules required to support it?

You'd have to also know which modules were loaded and what applications
they have registered.

I suspect doing so would take a partial loding of Asterisk. How is
apache's configtest implemented?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] problem transferring calls some of the times

[Ian]

Hi All

Agter a bit of logging to a syslog server, I found a peculiar entry 
today, ironically right after a call failed to transfer. They key 
sequence and call path used until it gets transferred is as follows


   * Phone rings on Asterisk
   * Asterisk transferres to the receptionists phone (GXP 2000)
 o Receptionist doensnt answer for 15 seconds, and the call
   gets routed to the bosses secrataries phone
   * Bosses secretary answers the phone and tries to transfer it to the
 boss with the keysequence "flash", extention "315", talks,
 "transfer" but the transfer is the one that fails

Message in the log is
Feb 22 09:55:22 10.219.127.102 GS_LOG: 
[00:0B:82:13:02:CF][000][FFFD][01010412] Received SIP message: 407
Feb 22 09:55:22 10.219.127.102 GS_LOG: 
[00:0B:82:13:02:CF][000][FFFD][01010412] SIP dialog matched to channel 0
Feb 22 09:55:22 10.219.127.102 GS_LOG: 
[00:0B:82:13:02:CF][000][FFFD][01010412] Send SIP message: ACK To 
10.219.127.7:5060, sip_handle: 0x0046F09C
Feb 22 09:55:22 10.219.127.102 GS_LOG: 
[00:0B:82:13:02:CF][000][FFFD][01010412] sip_len: 553, sip_handle: 
0x0046F09C, ACK sip:[EMAIL PROTECTED];user=p
hone SIP/2.0  Via: SIP/2.0/UDP 
10.219.127.102:5060;branch=z9hG4bK623e473ec5e8c5e8  From: "Wanda" 
;tag=1a7b934ecd3e23f7  To:
;tag=as07aa3c42  Contact: 
  Supported: 
path  Call-ID: 1138f5f7
[EMAIL PROTECTED]  CSeq: 58150 ACK  User-Agent: Grandstream 
BT200 1.1.4.18  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU

BSCRIBE,UPDATE,PRACK  Content-Length: 0


The message that got my worried is the one saying "Recieved SIP message 
407", can that be the ghost I am looking for?


Extentions.conf
[incoming_calls]
exten => s,1,NoOp(${CALLERID(name)} skakel Luzaan)
exten => s,n,dial(SIP/300,15)
exten => s,n,Set(CALLERID(name)=deur)
exten => s,n,Set(CALLERID(num)=deur)
exten => s,n,NoOp(${CALLERID(name)} skakel Wanda)
exten => s,n,dial(SIP/312)
;exten => s,n,dial(SIP/317)
exten => s,n,Hangup()

[internal]
exten => 900,1,Verbose(1|Echo test application)
exten => 900,n,Echo()
exten => 900,n,Hangup()
;interne oproepe

exten => _3XX,1,NoOp(${CALLERID} skakel ${EXTEN})
exten => _3XX,n,Dial(SIP/${EXTEN},30)
;exten => _3XX,n,execif(${CALLERID} != _3XX|goto|incoming_calls/s/1)
exten => _3XX,n,goto(incoming_calls,s,1)
exten => _3XX,n,Hangup()

Sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeers=yes
allowtransfer=yes
callevents=yes
regcontext=GXP_BLF

[sets](!)
type=friend
context=internal
host=dynamic
;disallow=all
;allow=speex
secret=test
dtmfmode=info
callgroup=1
pickupgroup=1
call-limit=20
subscribecontext=GXP_BLF
canreinvite=yes
nat=no

[300](sets) ;Luzaan
regexten=300

Any help will be apreciated

Thanks
Ian

Ian said the following on 22-Feb-08 10:06 AM:

Hi,

Mojo with Horan & Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call?  Meaning, for example, you transfer 
an incoming call to 700.  No problem.  Later, when it's picked up from 
701, can it NOT be transferred again? 


Moj
  

No I don't park the call.

The call comes in, and gets redirected to our receptionists phone, 
from there it gets transferred to another extension (the bosses 
secratary) and then gets transferred (to the boss). now the problem, 
sometimes that transfer fails, other times the call dont even want to 
leave the receptionists phone.


The big thing about this problem is that it comes and goes, like 
yesterday we didn't have a problem, and I did not change a thing.


Ian

Ian wrote:
  

Hi All

Sorry to be a bother again but seems like I just cant get away from 
the problems.


This time my problem is that *sometimes* a user cant transfer a call 
from one extension to another, I have narrowed down the problem to it 
only happening to calls from outside the internal system.


The wierd thing about the problem is that it comes and goes one moment 
the user can transfer, and the next call he can't.


I am running:

* Asterisk 1.4.17
* Zaptel 1.4.7.1
* Libpri 1.4.3

Using the following phones and firmware

* Grandstream GXP2000 (with ext pad) : 1.1.4.14
* Grandstream BT200 : 1.1.4.18

I have set up the phones to log debug logs to a syslog server, I am 
still trying to figure out what exactly the log says.


Is it an * problem, or Grandstream problem

Does anyone know if I am able to see the keysequence the user types 
into the phone (just in case it might even be a user made problem), I 
have tried scanning though the logs of a failed call, but could not 
see any lines that can be a keypress, or maybe I am looking in the 
incorrect spot?


Your help will be greatly appreciated.

Let me know if, in any way, I can shed some more light on the subject.

Thanks in advance
Ian
--
www.vddi.co.za 
I Coetzee
IT Tegnikus
Telefoon:   012 664 2300
Selfoon :   079 522 6519
Faks:   012 644 2902
E-pos   :   [EMAIL PR

Re: [asterisk-users] How to get a clean, basic configuration?

[Steve Langstaff]

> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tzafrir Cohen

> For the brave: use modules.conf without 'autoload = yes'. 
> This promises you many hours of interesting dialplan debugging. Enjoy.

Is there any method of automatically parsing a dialplan and generating a
list of the modules required to support it?

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Re: [asterisk-users] IAX: No outgoing audio for 10 seconds

[Tim H. Panton]

try setting 
transfer=no
or 
notransfer=yes
in iax.conf
Depending on the age of your asterisk version.

Tim.

- Original Message -
From: "randulo" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: 20 February 2008 18:12:01 o'clock (GMT) Europe/London
Subject: [asterisk-users] IAX: No outgoing audio for 10 seconds

I have an IAX hardphone connected to an Asterisk appliance sending and
receiving calls via IAX to three different providers. The appliance is
currently connected to a NAT router. The appliance is purposely being
set up via the GUI, not in messing with any config directly.
One of the "service providers"  is my other asterisk box on a
different Internet connection. In all cases, outgoing audio is silent
for about 10 seconds. If this were SIP, I'd immediately be suspicious
of NAT problems, but IAX? Ideas? I will be trying to hang the
appliance on the Internet directly but at the moment, this can't
happen.



-- 
Join our social networks+voip beta program : http://www.phonefromhere.com


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Re: [asterisk-users] problem transferring calls some of the times

[Ian]

Hi,

Mojo with Horan & Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call?  Meaning, for example, you transfer 
an incoming call to 700.  No problem.  Later, when it's picked up from 
701, can it NOT be transferred again? 


Moj
  

No I don't park the call.

The call comes in, and gets redirected to our receptionists phone, from 
there it gets transferred to another extension (the bosses secratary) 
and then gets transferred (to the boss). now the problem, sometimes that 
transfer fails, other times the call dont even want to leave the 
receptionists phone.


The big thing about this problem is that it comes and goes, like 
yesterday we didn't have a problem, and I did not change a thing.


Ian

Ian wrote:
  

Hi All

Sorry to be a bother again but seems like I just cant get away from 
the problems.


This time my problem is that *sometimes* a user cant transfer a call 
from one extension to another, I have narrowed down the problem to it 
only happening to calls from outside the internal system.


The wierd thing about the problem is that it comes and goes one moment 
the user can transfer, and the next call he can't.


I am running:

* Asterisk 1.4.17
* Zaptel 1.4.7.1
* Libpri 1.4.3

Using the following phones and firmware

* Grandstream GXP2000 (with ext pad) : 1.1.4.14
* Grandstream BT200 : 1.1.4.18

I have set up the phones to log debug logs to a syslog server, I am 
still trying to figure out what exactly the log says.


Is it an * problem, or Grandstream problem

Does anyone know if I am able to see the keysequence the user types 
into the phone (just in case it might even be a user made problem), I 
have tried scanning though the logs of a failed call, but could not 
see any lines that can be a keypress, or maybe I am looking in the 
incorrect spot?


Your help will be greatly appreciated.

Let me know if, in any way, I can shed some more light on the subject.

Thanks in advance
Ian
--
www.vddi.co.za 
I Coetzee
IT Tegnikus
Telefoon:   012 664 2300
Selfoon :   079 522 6519
Faks:   012 644 2902
E-pos   :   [EMAIL PROTECTED]
Skype   :   vddb_igcoetzee



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