Re: [asterisk-users] softphone instead of desktop phones

[Toufic Gmail]

No need for thirdlane or any proprietary extensions .
All you need is jssip open source that works with webrtc.
Just point it to wss of asterisk.
No restrictions nor min or max number and best of all it is free open source 
and web based (no software installation).
Also here is another web phone that can be used with asterisk 
https://tryit.jssip.net


You can find both on github





Sent from my iPhone
> On Apr 30, 2017, at 9:51 PM, Alex Epshteyn  wrote:
> 
> Thomas was asking how to save money and I was just offering an option. I am 
> sorry if my post was inappropriate.
> 
> That said, Thirdlane Connect itself is free, and we do offer a free version 
> for companies with up to 10 users. 
> 
> -- 
> 
> Alex Epshteyn
> email: a...@thirdlane.com
> web: www.thirdlane.com
> phone +1 415.261.6601
> 
> 
> - Original Message -
>> From: "Barry Flanagan" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Sunday, April 30, 2017 11:20:25 AM
>> Subject: Re: [asterisk-users] softphone instead of desktop phones
>> 
>> 
>> 
>> 
>> 
>> 
>> On 30 April 2017 at 16:54, Tech Support < aster...@voipbusiness.us >
>> wrote:
>> 
>> 
>> 
>> I thought this was a non-commercial list.
>> 
>> 
>> 
>> 
>> Yeah, I wouldn't mind so much if it had actually answered the
>> original poster's query. "Switch to our proprietary solution and we
>> can offer you this proprietary solution" isn't a contribution, it's
>> an ad.
>> 
>> 
>> -Barry
>> 
>> 
>> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto: asterisk-users-boun...@lists.digium.com ] On Behalf Of Alex
>> Epshteyn
>> Sent: Saturday, April 29, 2017 08:59 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] softphone instead of desktop phones
>> 
>> Thirdlane Connect can be used as a softphone. It works in modern
>> browsers
>> (no installation is required), on Mac, Windows and Linux desktops,
>> and on
>> mobile phones.
>> 
>> Besides basic softphone functionality, it provides instant messaging,
>> group
>> chat (channels), voice and video conferencing, and screen sharing. It
>> integrates with a variety of applications and CRMs such as
>> Salesforce, Zoho,
>> Zendesk, Redmine, etc.
>> 
>> Try it out!
>> 
>> 
>> --
>> 
>> Alex Epshteyn
>> web: www.thirdlane.com
>> 
>> 
>> - Original Message -
>>> From: "Amit Patkar" < a...@avhan.com >
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>> < asterisk-users@lists.digium.com >
>>> Sent: Saturday, April 29, 2017 9:16:05 AM
>>> Subject: Re: [asterisk-users] softphone instead of desktop phones
>>> 
>>> 
>>> Linphone is available for all major OS platforms.
>>> Then there is PortGo as well
>>> Regards,
>>> Amit Patkar
>>> 
>>> 
>>> On April 29, 2017 9:05:22 PM GMT+05:30, Thomas <
>>> thomasit...@gmail.com >
>>> wrote:
>>> 
>>> Hello,
>>> Iam lookong for an Softphone for iPhor oder Android smartphone
>>> using
>>> togehter with an headset.
>>> I tried Zoiper and CSipSimple but quality was bad compared to an
>>> desktop SIP phone.
>>> 
>>> Is there an better softphone?
>>> 
>>> Or are there softphone solutions for PC desktop MAC or Android with
>>> an
>>> headset?
>>> I want to save cost for desktop phones.
>>> 
>>> thanks Thomas
>>> 
>>> 
>>> --
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>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>>> --
>>> 
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>> 
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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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>> 
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>> 
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>> 
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>> 
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>> 
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Re: [asterisk-users] softphone instead of desktop phones

[Toufic Gmail]

Agree and that should be avoided. 

Sent from my iPhone

> On Apr 30, 2017, at 5:54 PM, Tech Support  wrote:
> 
> I thought this was a non-commercial list.
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Epshteyn
> Sent: Saturday, April 29, 2017 08:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] softphone instead of desktop phones
> 
> Thirdlane Connect can be used as a softphone. It works in modern browsers
> (no installation is required), on Mac, Windows and Linux desktops, and on
> mobile phones.
> 
> Besides basic softphone functionality, it provides instant messaging, group
> chat (channels), voice and video conferencing, and screen sharing. It
> integrates with a variety of applications and CRMs such as Salesforce, Zoho,
> Zendesk, Redmine, etc.
> 
> Try it out!
> 
> 
> -- 
> 
> Alex Epshteyn
> web: www.thirdlane.com
> 
> 
> - Original Message -
>> From: "Amit Patkar" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Saturday, April 29, 2017 9:16:05 AM
>> Subject: Re: [asterisk-users] softphone instead of desktop phones
>> 
>> 
>> Linphone is available for all major OS platforms.
>> Then there is PortGo as well
>> Regards,
>> Amit Patkar
>> 
>> 
>> On April 29, 2017 9:05:22 PM GMT+05:30, Thomas  
>> wrote:
>> 
>> Hello,
>> Iam lookong for an Softphone for iPhor oder Android smartphone using 
>> togehter with an headset.
>> I tried Zoiper and CSipSimple but quality was bad compared to an 
>> desktop SIP phone.
>> 
>> Is there an better softphone?
>> 
>> Or are there softphone solutions for PC desktop MAC or Android with an 
>> headset?
>> I want to save cost for desktop phones.
>> 
>> thanks Thomas
>> 
>> 
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> 
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>> 
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] softphone instead of desktop phones

[Toufic Gmail]

iPhone and android : growndwire 

Also you have media5 works well for iPhone

Linphone for iOS,android and Windows

Jitsi for windows works very well. 

Sent from my iPhone

> On Apr 29, 2017, at 5:35 PM, Thomas  wrote:
> 
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone using togehter 
> with an headset.
> I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP 
> phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with an 
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Hello Again - ooops

[aaberga/gmail]

Sorry forgot to attach the CLI trace:

=

CLI> pjsip show aors

  Aor:
Contact:  

 
=

  Aor:  210220

  Aor:  210320

  Aor:  messagenet_aor   0
Contact:  messagenet_aor/sip:sip.messagenet.it:5061  Unknown
   nan


-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' from AOR '2103' due 
to request
-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' from AOR '2103' due 
to request
-- Added contact 'sip:2103@37.228.255.229:60677;rinstance=635ece4650faa34e' 
to AOR '2103' with expiration of 900 seconds
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' from AOR '2102' due 
to request
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' from AOR '2102' due 
to request
-- Added contact 'sip:2102@37.228.255.229:60605;rinstance=fbd37b6a6d7cb4fb' 
to AOR '2102' with expiration of 900 seconds


-- Executing [2102@internal:1] Set("PJSIP/2103-0003", "ORIGIN=IP") in 
new stack
-- Executing [2102@internal:2] NoOp("PJSIP/2103-0003", "Declared 
CallerID=<"2103" <2103>>") in new stack
-- Executing [2102@internal:3] Set("PJSIP/2103-0003", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2102@internal:4] Set("PJSIP/2103-0003", 
"OriginalEXTEN=2102") in new stack
-- Executing [2102@internal:5] Set("PJSIP/2103-0003", 
"CDR(userfield)=2102") in new stack
-- Executing [2102@internal:6] Goto("PJSIP/2103-0003", 
"dialplan-switch,2102,1") in new stack
-- Goto (dialplan-switch,2102,1)
-- Executing [2102@dialplan-switch:1] NoOp("PJSIP/2103-0003", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2102@dialplan-switch:2] Dial("PJSIP/2103-0003", 
"PJSIP/2102") in new stack
[Sep 30 10:50:44] ERROR[19237]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2102'
[Sep 30 10:50:44] ERROR[19237]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2102'
[Sep 30 10:50:44] WARNING[19287][C-0003]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2102@dialplan-switch:3] Hangup("PJSIP/2103-0003", "") in 
new stack
  == Spawn extension (dialplan-switch, 2102, 3) exited non-zero on 
'PJSIP/2103-0003'
  
  
-- Executing [2103@internal:1] Set("PJSIP/2102-0004", "ORIGIN=IP") in 
new stack
-- Executing [2103@internal:2] NoOp("PJSIP/2102-0004", "Declared 
CallerID=<"2102" <2102>>") in new stack
-- Executing [2103@internal:3] Set("PJSIP/2102-0004", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2103@internal:4] Set("PJSIP/2102-0004", 
"OriginalEXTEN=2103") in new stack
-- Executing [2103@internal:5] Set("PJSIP/2102-0004", 
"CDR(userfield)=2103") in new stack
-- Executing [2103@internal:6] Goto("PJSIP/2102-0004", 
"dialplan-switch,2103,1") in new stack
-- Goto (dialplan-switch,2103,1)
-- Executing [2103@dialplan-switch:1] NoOp("PJSIP/2102-0004", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2103@dialplan-switch:2] Dial("PJSIP/2102-0004", 
"PJSIP/2103") in new stack
[Sep 30 10:52:01] ERROR[19299]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2103'
[Sep 30 10:52:01] ERROR[19299]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2103'
[Sep 30 10:52:01] WARNING[19306][C-0004]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2103@dialplan-switch:3] Hangup("PJSIP/2102-0004", "") in 
new stack
  == Spawn extension (dialplan-switch, 2103, 3) exited non-zero on 
'PJSIP/2102-0004'


=



Tnx,
Aldo


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[asterisk-users] Hello again

[aaberga/gmail]

Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with 
VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.



PJSIP seems tougher..

So my problem is that I do have a test system up in the cloud, behind a 
firewall. I am trying to make the “Hello World!” mandatory call between two 
iPhones (with the Bria SIP client).

Outcomes are erratic.



This is the pjsip.conf file:

——

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.2.12.3/32
local_net=127.0.0.1/32
external_media_address=10.2.12.2
external_signaling_address=10.2.12.2

;===Messagenet TRUNK 

[messagenet_reg]
type=registration
transport=transport-udp-nat
outbound_auth=messagenet_auth
server_uri=sip:xx...@sip.messagenet.it:5061
client_uri=sip:xx...@sip.messagenet.it:5061
 
[messagenet_auth]
type=auth
auth_type=userpass
password=
username=
 
[messagenet_aor]
type=aor
contact=sip:sip.messagenet.it:5061
 
[messagenet]
type=endpoint
transport=transport-udp-nat
context=messagenet_incoming
disallow=all
allow=ulaw
allow=alaw
outbound_auth=messagenet_auth
aors=messagenet_aor
 
[messagenet_id]
type=identify
endpoint=messagenet
match=sip.messagenet.it
 
;===Extension 2102

[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no


[auth2102]
type=auth
auth_type=userpass
password=xx
username=2102
 
[2102]
type=aor
max_contacts=1
 
;===Extension 2103

[2103]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2103
aors=2103
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no
 
[auth2103]
type=auth
auth_type=userpass
password=xx
username=2103
 
[2103]
type=aor
max_contacts=1
 


This is a trace of what I do see from the console.

First I let the Bria clients connect. Then I try to call terminal 1 from 
terminal 2. Most of the times there is no route to the destination, even if it 
appears as an online AOR (whatever that means!! Ahhh: Good olde times of Peer, 
Friend, etc… ;-)

A couple of times I got a connection, with the typical one side only audio of 
NAT traversal problems.

BTW: The iPhones are behind TWO nats (one is given by the broadband router, one 
by the WiFi router that gives a better WiFi cover for in-house things).

My understanding is that I did something wrong in letting the phones ‘register’ 
them as present and available to receive calls. 

If only I knew what is wrong… I have tried random combinations of 
rtp_symmetric, force_rport, and friends; nothing final discovered...



Thanks in advance for any help,
Aldo


PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP 
side. The only catch is that Zoiper has less than optimal background support on 
IOS… And I have no plan to make an IAX client myself!

I want to get my old Asterisk apps back online and the VOIP client part makes 
no sense to me..


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[asterisk-users] issue with CDR ODBC error on Ubuntu 15.04.1

[Toufic Khreish (Gmail)]

Hello,

 

I am trying to get CDR works for my asterisk installation.

My OS is Ubuntu 15 with asterisk 13.8 compiled locally on the machine.

MYSQL Server version is  5.6.28-0ubuntu0.15.04.1 (Ubuntu)

 

I also have another machine Ubuntu 15.04 same os but with asterisk 13.8.1
having the same issue, while same installation on Ubuntu 14.04

with 13.8.1 is working fine.

 

The only difference I saw is the mysql database engine version number on
Ubuntu 14.04 which was 5.5.

 

While there was no way to downgrade mysql to version 5.5 on Ubuntu 15.04 I
upgraded the mysql to version 5.6 on Ubuntu 14.04

surprisingly! the cdr kept on working. 

 

Would appreciate if someone can help solving this issue

 

 

The error that I am getting:

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:503 load_odbc_config: The
'pooling', 'shared_connections', 'limit', and 'idlecheck' options are
deprecated. Please see UPGRADE.txt for information

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:503 load_odbc_config: The
'pooling', 'shared_connections', 'limit', and 'idlecheck' options are
deprecated. Please see UPGRADE.txt for information

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:821 odbc_obj_connect:
res_odbc: Error SQLConnect=-1 errno=1045 [unixODBC]

[2016-04-15 19:24:34] NOTICE[1709]: res_odbc.c:585 load_odbc_config:
Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]

 

.

The command isql MySQL-asteriskcdrdb is working fine.

isql MySQL-asteriskcdrdb

+---+

| Connected!|

|   |

| sql-statement |

| help [tablename]  |

| quit  |

|   |

+---+

 

 

 

The following command returns errors : module reload cdr_adaptive_odbc.so

Module 'cdr_adaptive_odbc.so' reloaded successfully.

-- Reloading module 'cdr_adaptive_odbc.so' (Adaptive ODBC CDR backend)

  == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': Found

[2016-04-15 19:31:41] WARNING[1758]: cdr_adaptive_odbc.c:135 load_config: No
such connection 'MySQL-asteriskcdrdb' in the 'asteriskcdrdb' section of
cdr_adaptive_odbc.conf.  Check res_odbc.conf.

 

 

odbc show,  returns the following

 

ODBC DSN Settings

-

 

  Name:   asteriskcdrdb

  DSN:MySQL-asteriskcdrdb

Last connection attempt: 2016-04-15 19:24:40

 

 

 

cdr show status

 

Call Detail Record (CDR) settings

--

  Logging:Enabled

  Mode:   Simple

  Log unanswered calls:   No

  Log congestion: No

 

* Registered Backends

  ---

Adaptive ODBC

cdr_manager

 

 

 

 

My ODBC related files:

1.   cdr_adaptive_odbc.conf

[asteriskcdrdb]

connection=MySQL-asteriskcdrdb

loguniqueid=yes

table=cdr

alias start => calldate

 

2.   odbcinst.ini

[MySQL]

Description = ODBC for MySQL

Driver = /usr/lib/arm-linux/odbc/libmyodbc.so

Setup = /usr/lib/arm-linux/odbc/libodbcmyS.so

FileUsage = 1

polling=no

 

3.   odbc.ini

[MySQL-asteriskcdrdb]

Description=MySQL connection to 'asteriskcdrdb' database

driver=MySQL

server=localhost

Port=3306

username=asterisk

password=xx

Socket=/run/mysqld/mysqld.sock

option=3

database=asteriskcdrdb

 

 

4.   res_odbc_additional.conf

[asteriskcdrdb]

enabled=>yes

dsn=>MySQL-asteriskcdrdb

pooling=>no

limit=>1

pre-connect=>yes

username=>asterisk

password=>xx

database=>asteriskcdrdb

 

5.   cel_odbc_custom.conf

[cel]

connection=MySQL-asteriskcdrdb

loguniqueid=yes

table=cel

 

Thank you in advance.

 

 

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Re: [asterisk-users] CDR ODBC error

[Toufic Khreish (Gmail)]

Hello,

 

I am trying to get CDR work for my asterisk installation.

My OS is Ubuntu 15 with asterisk 13.8 compiled locally on the machine.

MYSQL Server version is  5.6.28-0ubuntu0.15.04.1 (Ubuntu)

 

Would appreciate if someone can help solving this issue

 

 

The error that I am getting:

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:503 load_odbc_config: The 
'pooling', 'shared_connections', 'limit', and 'idlecheck' options are 
deprecated. Please see UPGRADE.txt for information

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:503 load_odbc_config: The 
'pooling', 'shared_connections', 'limit', and 'idlecheck' options are 
deprecated. Please see UPGRADE.txt for information

[2016-04-15 19:24:34] WARNING[1709]: res_odbc.c:821 odbc_obj_connect: res_odbc: 
Error SQLConnect=-1 errno=1045 [unixODBC]

[2016-04-15 19:24:34] NOTICE[1709]: res_odbc.c:585 load_odbc_config: Registered 
ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]

 

.

The command isql MySQL-asteriskcdrdb is working fine.

isql MySQL-asteriskcdrdb

+---+

| Connected!|

|   |

| sql-statement |

| help [tablename]  |

| quit  |

|   |

+---+

 

Note that could help . once on the sql command line I have to issue : 

1.   use asteriskcdrdb 

then 

2.   select * from cdr

I cannot issue first : select * from cdr   it does not work. Could be that my 
connection MySQL-asteriskcdrdb is not sending the database name along ? 

 

 

The following command returns errors : module reload cdr_adaptive_odbc.so

Module 'cdr_adaptive_odbc.so' reloaded successfully.

-- Reloading module 'cdr_adaptive_odbc.so' (Adaptive ODBC CDR backend)

  == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': Found

[2016-04-15 19:31:41] WARNING[1758]: cdr_adaptive_odbc.c:135 load_config: No 
such connection 'MySQL-asteriskcdrdb' in the 'asteriskcdrdb' section of 
cdr_adaptive_odbc.conf.  Check res_odbc.conf.

 

 

odbc show,  returns the following

 

ODBC DSN Settings

-

 

  Name:   asteriskcdrdb

  DSN:MySQL-asteriskcdrdb

Last connection attempt: 2016-04-15 19:24:40

 

 

 

cdr show status

 

Call Detail Record (CDR) settings

--

  Logging:Enabled

  Mode:   Simple

  Log unanswered calls:   No

  Log congestion: No

 

* Registered Backends

  ---

Adaptive ODBC

cdr_manager

 

 

 

 

My ODBC related files:

1.   cdr_adaptive_odbc.conf

[asteriskcdrdb]

connection=MySQL-asteriskcdrdb

loguniqueid=yes

table=cdr

alias start => calldate

 

2.   odbcinst.ini

[MySQL]

Description = ODBC for MySQL

Driver = /usr/lib/arm-linux/odbc/libmyodbc.so

Setup = /usr/lib/arm-linux/odbc/libodbcmyS.so

FileUsage = 1

polling=no

 

3.   odbc.ini

[MySQL-asteriskcdrdb]

Description=MySQL connection to 'asteriskcdrdb' database

driver=MySQL

server=localhost

Port=3306

username=asterisk

password=xx

Socket=/run/mysqld/mysqld.sock

option=3

database=asteriskcdrdb

 

 

4.   res_odbc_additional.conf

[asteriskcdrdb]

enabled=>yes

dsn=>MySQL-asteriskcdrdb

pooling=>no

limit=>1

pre-connect=>yes

username=>asterisk

password=>xx

database=>asteriskcdrdb

 

5.   cel_odbc_custom.conf

[cel]

connection=MySQL-asteriskcdrdb

loguniqueid=yes

table=cel

 

Thank you in advance.

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Thursday, February 11, 2016 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] CDR ODBC error

 

 

 

On Tue, Feb 9, 2016 at 4:39 PM, Carlos Chavez  > wrote:

I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep 
getting this error:

[Feb  9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in 
ExecDirect: -1, query is: INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence)
 VALUES ({ts '2016-02-09 16:21:28'},?,?,?,?,?,?,?,?,?,?,?,?,?,?,? ,?,?,?)
[Feb  9 16:21:43] WARNING[2088]: res_odbc.c:612 ast_odbc_direct_execute: SQL 
Execute error! Verifying connection to asterisk [asterisk]...
[Feb  9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in 
ExecDirect: -1, query is: INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence)
 VALUES ({ts '2016-02-09 

Re: [asterisk-users] tls on asterisk 13

[Toufic Gmail]

I did using acrobits groundwire on asterisk 13.7.2
Had to add a statement in pjsip.endpointxxx
I do not have it in mind but can look it up for you tomorrow. 

Sent from my iPhone

> On Jul 8, 2015, at 9:05 PM, ricky gutierrez  wrote:
> 
> Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed
> to make it work, all my terminals spa Cisco 5XX
> 
> look my cli
> 
> [Jul  8 11:09:16] ERROR[14733]: pjsip:0 :tlsc0x7f539801 TLS
> connect() error: Connection refused [code=120111]
> [Jul  8 11:09:16] WARNING[14733]: pjsip:0 :  tsx0x7f53a8008 Failed
> to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!
> err=120111 (Connection refused)
> [Jul  8 11:09:46] ERROR[14733]: pjsip:0 :tlsc0x7f539801 TLS
> connect() error: Connection refused [code=120111]
> [Jul  8 11:09:46] WARNING[14733]: pjsip:0 :  tsx0x7f53a8008 Failed
> to send Request msg OPTIONS/cseq=31917 (tdta0x7f53c000dcb0)!
> err=120111 (Connection refused)
> 
> someone has had good results with tls
> 
> my config
> [transport-tls]
> type=transport
> protocol=tls
> bind=0.0.0.0:5061
> cert_file=/etc/asterisk/keys/asterisk.crt
> priv_key_file=/etc/asterisk/keys/asterisk.key
> method=tlsv1
> 
> []
> type=endpoint
> context=XX-Xip
> disallow=all
> allow=ulaw
> allow=alaw
> transport=transport-tls
> direct_media=no
> force_rport=yes
> rtp_symmetric=yes
> mailboxes=@default
> auth=
> aors=
> media_encryption=sdes
> dtmfmode=rfc4733
> 
> 
> regardss
> 
> -- 
> rickygm
> 
> http://gnuforever.homelinux.com
> 
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Toufic Gmail]

Please set correct MTU at server side, it is definitely an MTU issue.  

Sent from my iPhone

> On Mar 3, 2016, at 5:31 PM, Olivier  wrote:
> 
> Hello,
> 
> I'm remotely managing an asterisk setup using an OpenVPN client on this 
> Asterisk box, connecting to an OpenVPN server of mine).
> 
> This box is mainly connected to PSTN.
> It is also connected to the Internet, only for remote management.
> 
> The former ADSL link has recently been replaced by a new 4G link (UMTS).
> 
> I'm connecting to this box from a Debian Jessie/Gnome Terminal combo.
> 
> With this new link, whenever I launch a vim, a nano or a rasterisk session, 
> my terminal freezes (rasterisk) or remains empty (nano, vim).
> 
> When a session is frozon, I can open a new one at the same so it excludes a 
> basic connectivity loss.
> 
> What would you suggest ?
> 
> Best regards
> 
> 
> PS: I was about to determine best MTU value but I always thought a punishment 
> for a bad MTU value would be a lower throughput, not a screen freeze. Is it 
> correct ?
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Re: [asterisk-users] Product CDR/Queue/Meetme

[Toufic Khreish (Gmail)]

Hello,

 

I am interested.

 

Regards

Toufic

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helvio Junior
Sent: Monday, June 22, 2015 5:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Product CDR/Queue/Meetme

 

Gentleman,

Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me.

I had some difficult looking for a Asterisk software that provide me some
functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i
decided to develop than.

In a few weeks i'll deploy a Beta version of this software and i'd like to
know if is somebody available to try this beta and free version?

If you don't want to try this version but would like to see/suggest any
feature in this software, let me know.

Forecast functions to Beta Version:

*   Realtime view for:

*   Queues;
*   Peers (Similar as BLF);
*   Trunk calls/utilization;

*   MeetMe

*   Create, modify, delete and schedule;
*   Real time view of members;
*   Delete members;
*   Mute/Unmute;
*   Send Invite by e-mail (with .VCS file)

*   Dialer

*   Create dialer (by campaign with contacts) 
*   Monitoring of campaig, calls, and status; 
*   Time control to retry failed call
*   Control of day time to call (commercial time, full time, etc...)

*   Charts and reports:

*   Trunk utilization;
*   CDR;
*   Queues (Most common reports and charts, distributions, times,
etc...)
*   Export to Excel Spreadsheet and PDF File
*   Report Scheduler
*   Much more...

*   REST API for 100% of functionalities;
*   Admin and User Console 100% Web HTML5;
*   Developed in Windows with C#;
*   Integrate with Asterisk using AMI only;
*   Allow manage many Asterisk that you want using same instance of this
software (One software and one installation);


Obs.: I'll provide a Full License for everybody that help me trying the Beta
version.



-- 
 
Att,
Hélvio Junior
SafeId - Gestão de identidades e Acessos
+55 41 | 9893-2694, single-sign-on.com.br
helvio.jun...@safetrend.com.br mailto:helvio.jun...@safetrend.com.br 
-- 
 
Att,
Hélvio Junior
SafeId - Gestão de identidades e Acessos
+55 41 | 9893-2694, single-sign-on.com.br
helvio.jun...@safetrend.com.br mailto:helvio.jun...@safetrend.com.br 
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Re: [asterisk-users] Seeking advice about ISDN BRI Cards

[Toufic Khreish (Gmail)]

Beronet Gateway BFSB2HY , it works well for me two.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markos Vakondios
Sent: Tuesday, May 26, 2015 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seeking advice about ISDN BRI Cards

 

Checkout Beronet ISDN cards and Berofix Gateway (appliance or pci card)

 

Personally for my last installation I chose Berofix card which is rock solid 
and reliable, yet easily configurable.

 

With berofix you don't need telephony drivers on the host system, the isdn card 
is detected as a NIC and all configuration is done using a web interface.

 

Then you configure FreePBX a new trunk to use the berofix IP and that's it!

 

This way the mISDN and other channel drivers' burden is skipped and works very 
well for me so far!

 

 

 

 

 

On 26 May 2015 at 16:52, Olivier oza.4...@gmail.com 
mailto:oza.4...@gmail.com  wrote:

 

 

2015-05-26 12:17 GMT+02:00 Lukasz Sokol el.es...@gmail.com 
mailto:el.es...@gmail.com :

Hi,
please whoever has some expertise in choice of BRI ISDN cards,
please restore my faith in community support :)

(on private email I can probably explain more than fits for a public forum)

Most I'd like to ask is about what to choose, out of what is available...

My locality is United Kingdom, lines from British Telecom (BT),
but any advice / pointers (I googled around already) are welcome...

My system to fit this card into, is FreePBX Distro with Asterisk 11,
already running with incoming SIP trunk(s);
I wish to extend it to accept incoming 'landline' ISDN BRI (6 channels / 3 
ports).

So far the interesting option(s) were Sangoma A500 and Digium B410P...
(the appliance is adopted from an old desktop that still only ever has PCI2.0 
slots,
no PCIE)

 

I would suggest to also consider Digium Hx8  boards which exist in PCI format.


[1] http://www.digium.com/en/products/telephony-cards/hybrid/h8

(there are also OpenVOX's cards, although their installation guide is somewhat, 
well...
 in the old kernel era...)

Anyone who use(d) any of the above, not necessarily on a FreePBX - you're 
welcome... :)

Kind Regards,
Lukasz


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Re: [asterisk-users] WEBRTC is no longer working with Firefox after upgrade to version 37

[Toufic Khreish (Gmail)]

Thank you.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, April 08, 2015 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] WEBRTC is no longer working with Firefox after
upgrade to version 37

Toufic Khreish (Gmail) wrote:
 Hello,

 Webrtc stopped after upgrading firefox from version 36 to version37.
 I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 
 and firefox version 36 without any issues until firefox was upgraded 
 to version 37.
 Unfortunately Chrome works well in one direction (from chrome to any
 extension) but calling from an extension to a webrtc on chrome has one 
 way voice.

 Could someone try to investigate the problem of firefox version37.0.1 
 with webrtc ? no voice in any direction.
 Should we try it with a computer that has not an updated version of 
 firefox things work normally, also if we rollback (install version 36, 
 it works
 well)

Someone already filed an Asterisk issue[1] and there is also a Firefox
issue[2]. It's also been fixed in Firefox 38 already.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-24911
[2] https://bugzilla.mozilla.org/show_bug.cgi?id=1147919

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
www.digium.com  www.asterisk.org

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[asterisk-users] WEBRTC is no longer working with Firefox after upgrade to version 37

[Toufic Khreish (Gmail)]

Hello,

Webrtc stopped after upgrading firefox from version 36 to version37.
I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and
firefox version 36 without any issues until firefox was upgraded to version
37.
Unfortunately Chrome works well in one direction (from chrome to any
extension) but calling from an extension to a webrtc on chrome has one way
voice.

Could someone try to investigate the problem of firefox version37.0.1 with
webrtc ? no voice in any direction.
Should we try it with a computer that has not an updated version of firefox
things work normally, also if we rollback (install version 36, it works
well)

Thank you and best regards
Toufic KHREISH



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Re: [asterisk-users] Connecting Samsung Galaxy to Asterisk for VoLTE

[Toufic Khreish (Gmail)]

Hi,

 

I have tried Groundwire on IOS , and Android Alcatel (voice and video calls 
with asterisk 13.3)

Also tried Bria on both OS in video and voice.

 

Regards

Toufic

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sevana Oy
Sent: Friday, April 03, 2015 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connecting Samsung Galaxy to Asterisk for VoLTE

 

Hi,

Has anybody tried to connect Samsung Galaxy to Asterisk PBX to be able to make 
calls over VoLTE?

Thanks a lot in advance!

Best regards,

Sevana

http://www.sevana.biz

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[asterisk-users] Asterisk 13.3.0 IAX trunk issue with Yeastar

[Toufic Khreish (Gmail)]

Hello,

I have a weird problem between Asterisk 13.3 and a Yeastar U200 pbx over IAX
trunk.
Should I call from Yeastar to my asterisk 13.3 the call goes through without
issues.
Should I call from asterisk 13.3 to Yeastar I can hear a ring tone however
the yeastar does not show any activities.
On the yeastar I initiated a debug commandiax2 set debug peer my
trunk name 
While I hear the ring from my side nothing appears in the debug of Yeastar
pbx.

On the asterisk 13.3 debug terminal I see that the call was initiated .


Same setting is working between an asterisk 13.2 and the Yeastar.
Can anyone help ?

I will try IAX trunk between Asterisk 13.3 and asterisk 13.2 to check if it
works.

Regards
Toufic


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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

Hello Matthew,

The asterisk crashing issue was solved with the Asterisk 13.3.0, now video
calls are okay between all devices.
The only issue left is with the Grandstream GXV3175 where video is still
very slow (downstream), it shows on the LCD 1 frame every few seconds.

Hope this helps and should someone has a suggestion on how to solve the
GXV3175 video would be great.

Best regards
Toufic

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Wednesday, March 18, 2015 4:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I see that my asterisk is started with the -g option, the core file I 
 cannot find on my system (find / -name core*)


I would suspect one of the following:

(1) Asterisk is not actually crashing.
(2) Something is deleting the core files.
(3) The core files are hiding really, really well.

Either way, if you can't get a backtrace, there isn't much we can do to help
with that problem.

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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

I can assure you that asterisk is crashing, as when I try to reconnect I see
it reloading again.
Could be that something is deleting the core ! is there a way to find the
path to where the core files are stored?
My system is Lubuntu ,  Linux #41 SMP PREEMPT Tue Nov 11 16:35:58 CST 2014
armv7l armv7l armv7l GNU/Linux
Operating systemUbuntu Linux 14.04.1

---
Toufic KHREISH

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Wednesday, March 18, 2015 4:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I see that my asterisk is started with the -g option, the core file I 
 cannot find on my system (find / -name core*)


I would suspect one of the following:

(1) Asterisk is not actually crashing.
(2) Something is deleting the core files.
(3) The core files are hiding really, really well.

Either way, if you can't get a backtrace, there isn't much we can do to help
with that problem.

--
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

Attached is my safe_asterisk script, it is moving the core to some dumpdrop
directory that does not seem to exist.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Wednesday, March 18, 2015 1:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

If you take a look at the safe_asterisk shell script, usually located at
/usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where
the core files are located. If it's not located there, then you'll need to
look at the Asterisk init script for the scripts location. I hope this
helps.
Regards;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 17, 2015 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I see that my asterisk is started with the -g option, the core file I 
 cannot find on my system (find / -name core*)


I would suspect one of the following:

(1) Asterisk is not actually crashing.
(2) Something is deleting the core files.
(3) The core files are hiding really, really well.

Either way, if you can't get a backtrace, there isn't much we can do to help
with that problem.

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
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#!/bin/sh

ASTETCDIR=/etc/asterisk
ASTSBINDIR=/usr/sbin
ASTVARRUNDIR=/var/run/asterisk
ASTVARLOGDIR=/var/log/asterisk

CLIARGS=$*# Grab any args passed to safe_asterisk
TTY=9   # TTY (if you want one) for Asterisk to run on
CONSOLE=yes # Whether or not you want a console
#NOTIFY=root@localhost  # Who to notify about crashes
#EXEC=/path/to/somescript   # Run this command if Asterisk crashes
#LOGFILE=${ASTVARLOGDIR}/safe_asterisk.log# Where to place the normal 
logfile (disabled if blank)
#SYSLOG=local0  # Which syslog facility to use (disabled if 
blank)
MACHINE=`hostname`  # To specify which machine has crashed when 
getting the mail
DUMPDROP=${DUMPDROP:-/tmp}
RUNDIR=${RUNDIR:-/tmp}
SLEEPSECS=4
ASTPIDFILE=${ASTVARRUNDIR}/asterisk.pid

# comment this line out to have this script _not_ kill all mpg123 processes when
# asterisk exits
KILLALLMPG123=1

# run asterisk with this priority
PRIORITY=0

# set system filemax on supported OSes if this variable is set
# SYSMAXFILES=262144

# Asterisk allows full permissions by default, so set a umask, if you want
# restricted permissions.
#UMASK=022

# set max files open with ulimit. On linux systems, this will be automatically
# set to the system's maximum files open devided by two, if not set here.
# MAXFILES=32768

message() {
if test -n $TTY  test $TTY != no; then
echo $1 /dev/${TTY}
fi
if test -n $SYSLOG; then
logger -p ${SYSLOG}.warn -t safe_asterisk[$$] $1
fi
if test -n $LOGFILE; then
echo safe_asterisk[$$]: $1 $LOGFILE
fi
}

# Check if Asterisk is already running.  If it is, then bug out, because
# starting safe_asterisk when Asterisk is running is very bad.
VERSION=`${ASTSBINDIR}/asterisk -nrx 'core show version' 2/dev/null`
if test `echo $VERSION | cut -c 1-8` = Asterisk; then
message Asterisk is already running.  $0 will exit now.
exit 1
fi

# since we're going to change priority and open files limits, we need to be
# root. if running asterisk as other users, pass that to asterisk on the command
# line.
# if we're not root, fall back to standard everything.
if test `id -u` != 0; then
echo Oops. I'm not root. Falling back to standard prio and file max. 
2
echo This is NOT suitable for large systems. 2
PRIORITY=0
message safe_asterisk was started by `id -n` (uid `id -u`).
else

Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

I see that my asterisk is started with the -g option, the core file I cannot
find on my system (find / -name core*)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 17, 2015 1:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Mon, Mar 16, 2015 at 6:12 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 Hello Matthew,

 I have compiled Asterisk 13.2 with the following compiler Flags enabled:
 DON'T_OPTIMIZE
 DEBUG THREADS
 BETTER_BACKTRACES


 My asterisk is running with the asterisk_script:
 root 24048 39.4  2.4 128564 50640 pts/1Sl   00:02   2:21
 /usr/sbin/asterisk -f -vvvg -c

 core show locks

 ==
 =
 === 13.2.0
 === Currently Held Locks
 ==
 =
 ===
 === pending lock# (file): lock type line num function 
 lock
 name lock addr (times locked)
 ===
 ==
 =

 When my asterisk crashes there is no file called core.

 The results of  gdb -se asterisk -ex bt full -ex thread apply all bt
 --batch -c core  /tmp/backtrace.txt

 /usr/src/asterisk-13.2.0/core: No such file or directory.
 No stack.

 What could be the problem ?


(1) Asterisk only generates a core file if started with the '-g' option

(2) Your core file may not be located in the directory that you are running
gdb from. You will need to find where the core file was located - this is
typically determined by /proc/sys/kernel/core_pattern

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

Hello Matthew,

I have compiled Asterisk 13.2 with the following compiler Flags enabled:
DON'T_OPTIMIZE
DEBUG THREADS
BETTER_BACKTRACES


My asterisk is running with the asterisk_script:
root 24048 39.4  2.4 128564 50640 pts/1Sl   00:02   2:21
/usr/sbin/asterisk -f -vvvg -c

core show locks

===
=== 13.2.0
=== Currently Held Locks
===
===
=== pending lock# (file): lock type line num function lock
name lock addr (times locked)
===
===

When my asterisk crashes there is no file called core.

The results of  gdb -se asterisk -ex bt full -ex thread apply all bt
--batch -c core  /tmp/backtrace.txt

/usr/src/asterisk-13.2.0/core: No such file or directory.
No stack.

What could be the problem ?

Best regards
Toufic
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Thursday, March 12, 2015 3:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 Thank you, I needed a starting point to start my post.

 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
 Voice issues on IAX2 Trunks, All extensions are SIP.
 The IAX2 trunks on Asterisk 12.8.1 produces only  one error out of : 
 iax2 set debug trunk on
 [2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793
 compress_subclass: Can't compress subclass 2097217

 On the box running asterisk 1.6.2.6 I receive the following warning:
 [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no 
 samples for alawtolin


 core show channels
 Channel  Location State   Application(Data)
 IAX2/Mypbx1-15288(None)   Up  AppDial((Outgoing Line))
 SIP/6000-000f(None)   Up
 Dial(IAX2/Mypbx1/300,300,Tt)
 2 active channels

 Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw 
 and GSM codecs) Voice is not very clear and choppy

 If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 
 , voice is very clear.

Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm
going to skip past this issue.

 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues).

 Calls from Bria video sip phone (android or IOS) to Grandstream 
 GXV3175 (asterisk engine stops/crashes)

Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1]
and file an issue on the issue tracker [2]. A pcap of the message traffic
would also be very helpful.

 Call from Groundwire video sip (IOS since Android version does not 
 H264
 codec) to Grandstream GXV3175, Asterisk stops

I'm going to assume Asterisk stops means it crashed as well. If you'd like
to get a backtrace for that as well and attach it to the same issue, that
would be helpful - it may be the same problem that you see with the Bria
phone, or it may be something else.

 Calls between SIP Video softphones works well no issues.

Well, that's good. :-)

 Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well.
 (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) 
 Calls between GXV3275 and GXV3175 video streaming is very slow on the
 GXV3175 (this is not the case under Asterisk 12.8.1) Calls from 
 GXV3175 to Bria (video is displayed on bria side only)

Since there are some that work fine, and some that don't, the trick is going
to be knowing:
(1) How the SIP peers (or PJSIP endpoints) are configured
(2) How the phones are negotiating media with Asterisk

Both your SIP configuration as well as a DEBUG log - generated with trace
logging, showing the negotiation [3] - will be needed to figure out what is
occurring.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira/
[3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing

Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

I will rebuild my asterisk with the options enabled ONT_OPTIMIZE and
BETTER_BACKTRACES
Then I will create the traces and post them as per your recommendations.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Thursday, March 12, 2015 3:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 Thank you, I needed a starting point to start my post.

 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
 Voice issues on IAX2 Trunks, All extensions are SIP.
 The IAX2 trunks on Asterisk 12.8.1 produces only  one error out of : 
 iax2 set debug trunk on
 [2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793
 compress_subclass: Can't compress subclass 2097217

 On the box running asterisk 1.6.2.6 I receive the following warning:
 [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no 
 samples for alawtolin


 core show channels
 Channel  Location State   Application(Data)
 IAX2/Mypbx1-15288(None)   Up  AppDial((Outgoing Line))
 SIP/6000-000f(None)   Up
 Dial(IAX2/Mypbx1/300,300,Tt)
 2 active channels

 Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw 
 and GSM codecs) Voice is not very clear and choppy

 If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 
 , voice is very clear.

Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm
going to skip past this issue.

 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues).

 Calls from Bria video sip phone (android or IOS) to Grandstream 
 GXV3175 (asterisk engine stops/crashes)

Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1]
and file an issue on the issue tracker [2]. A pcap of the message traffic
would also be very helpful.

 Call from Groundwire video sip (IOS since Android version does not 
 H264
 codec) to Grandstream GXV3175, Asterisk stops

I'm going to assume Asterisk stops means it crashed as well. If you'd like
to get a backtrace for that as well and attach it to the same issue, that
would be helpful - it may be the same problem that you see with the Bria
phone, or it may be something else.

 Calls between SIP Video softphones works well no issues.

Well, that's good. :-)

 Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well.
 (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) 
 Calls between GXV3275 and GXV3175 video streaming is very slow on the
 GXV3175 (this is not the case under Asterisk 12.8.1) Calls from 
 GXV3175 to Bria (video is displayed on bria side only)

Since there are some that work fine, and some that don't, the trick is going
to be knowing:
(1) How the SIP peers (or PJSIP endpoints) are configured
(2) How the phones are negotiating media with Asterisk

Both your SIP configuration as well as a DEBUG log - generated with trace
logging, showing the negotiation [3] - will be needed to figure out what is
occurring.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira/
[3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
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   http://www.asterisk.org/hello

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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

Should I unload or rename the res_format_attr_h264.soH.264 Format
Attribute Module
The asterisk server 13.2.0 does not break anymore upon calls towards GXV3175
grandstream, however only downstream video displayed on the GXV3175 is very
slow (1 frame per 10 seconds)

This problem only concerns GXV3175 for the moment (with the
res_format_attr_h264.so removed). (GXV3175 version  Hardware : 1.4A ,
program version: 1.0.3.76 and CPE version 1.0.1.32)

Any idea why ? and how could this be fixed ?


-Original Message-
From: Toufic Khreish (Gmail) [mailto:toufic.khre...@gmail.com] 
Sent: Tuesday, March 10, 2015 11:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk 13.2.0 Video issues

Thank you, I needed a starting point to start my post.

1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
Voice issues on IAX2 Trunks, All extensions are SIP.
The IAX2 trunks on Asterisk 12.8.1 produces only  one error out of : iax2
set debug trunk on
[2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793
compress_subclass: Can't compress subclass 2097217

On the box running asterisk 1.6.2.6 I receive the following warning:
[2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples
for alawtolin


core show channels
Channel  Location State   Application(Data)
IAX2/Mypbx1-15288(None)   Up  AppDial((Outgoing Line))
SIP/6000-000f(None)   Up
Dial(IAX2/Mypbx1/300,300,Tt)
2 active channels

Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and
GSM codecs) Voice is not very clear and choppy 

If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 ,
voice is very clear.

2. Asterisk 13.2.0 Video issues (no IAX2 voice issues).

Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175
(asterisk engine stops/crashes) Call from Groundwire video sip (IOS since
Android version does not H264 codec) to Grandstream GXV3175, Asterisk stops

Calls between SIP Video softphones works well no issues.
Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well.
(Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) Calls
between GXV3275 and GXV3175 video streaming is very slow on the GXV3175
(this is not the case under Asterisk 12.8.1) Calls from GXV3175 to Bria
(video is displayed on bria side only)

There might be an issue on the Grandstream sip video phones as far as H264
is concerned however the case of streaming slowness is not there under
Asterisk 12.8.1) I cannot find anything related to the moment where asterisk
is breaking upon calling GXV3175

Best regards
Khreish Toufic


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 10, 2015 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting 
 problems with the format H264, Asterisk 12.8.1 compiled on the same 
 hardware is behaving very well for the same format H264

 Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality.

 Could someone investigate the problem of Asterisk 13 with video 
 support on
 H264 ?


There's no where near enough information in your e-mail to give someone an
indication on where to start.

What channels are involved? What are their configurations? What formats are
negotiated on the channels? What symptoms do you see? What does the CLI
show, both when active calls are running and for a 'core show channel' for
the involved parties?

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

Thank you, I needed a starting point to start my post.

1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
Voice issues on IAX2 Trunks, All extensions are SIP.
The IAX2 trunks on Asterisk 12.8.1 produces only  one error out of : iax2
set debug trunk on 
[2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793
compress_subclass: Can't compress subclass 2097217

On the box running asterisk 1.6.2.6 I receive the following warning:
[2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples
for alawtolin


core show channels
Channel  Location State   Application(Data)
IAX2/Mypbx1-15288(None)   Up  AppDial((Outgoing Line))
SIP/6000-000f(None)   Up
Dial(IAX2/Mypbx1/300,300,Tt)
2 active channels

Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and
GSM codecs)
Voice is not very clear and choppy 

If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 ,
voice is very clear.

2. Asterisk 13.2.0 Video issues (no IAX2 voice issues).

Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175
(asterisk engine stops/crashes)
Call from Groundwire video sip (IOS since Android version does not H264
codec) to Grandstream GXV3175, Asterisk stops

Calls between SIP Video softphones works well no issues.
Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well.
(Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish)
Calls between GXV3275 and GXV3175 video streaming is very slow on the
GXV3175 (this is not the case under Asterisk 12.8.1)
Calls from GXV3175 to Bria (video is displayed on bria side only)

There might be an issue on the Grandstream sip video phones as far as H264
is concerned however the case of streaming slowness is not there under
Asterisk 12.8.1)
I cannot find anything related to the moment where asterisk is breaking upon
calling GXV3175

Best regards
Khreish Toufic


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 10, 2015 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting 
 problems with the format H264, Asterisk 12.8.1 compiled on the same 
 hardware is behaving very well for the same format H264

 Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality.

 Could someone investigate the problem of Asterisk 13 with video 
 support on
 H264 ?


There's no where near enough information in your e-mail to give someone an
indication on where to start.

What channels are involved? What are their configurations? What formats are
negotiated on the channels? What symptoms do you see? What does the CLI
show, both when active calls are running and for a 'core show channel' for
the involved parties?

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 13.2.0 Video issues

[Toufic Khreish (Gmail)]

I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems
with the format H264, Asterisk 12.8.1 compiled on the same hardware is
behaving very well for the same format H264

Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality.

Could someone investigate the problem of Asterisk 13 with video support on
H264 ?

Thank you. 

 

 

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Re: [asterisk-users] Asterisk - WiMax Island Use

[Ochere Gmail]

It can be done, contact me offlist to discuss further

Frank

Sent from my iPhone

 On Jan 14, 2015, at 7:32 PM, j.halif...@seznam.cz j.halif...@seznam.cz 
 wrote:
 
 Hello All,
 
 Please advise kindly about the following arrangement:
 
 I need to have Asterisk working with company's mobiles via company's WiMax 
 mobile network. Both Asterisk and WiMax can work in an island mode (i.e. not 
 necessarily connected, even preferably not connected to any other 
 communication network like mobile operator, Intranet or Internet).
 
 The result desired should be that company's field workers can communicate 
 with each other by means of company's mobile phones, company's WiMax 
 technology and company's Asterisk, but preferably not with any mobile 
 operator's subscribers.
 
 Please advise kindly whether such arrangement is possible and if so, what 
 should I study to know which devices to buy.etc.
 
 Thank you for your great help. :)
 
 BR,
 JH
 
 =
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Re: [asterisk-users] Digium Analog card and Asterisk

[Andre Courchesne - Gmail]

Or educate him ! 

Sent from my iPhone

On 2013-07-06, at 3:03 PM, Steve Edwards asterisk@sedwards.com wrote:

 On Sat, 6 Jul 2013, William Muriithi wrote:
 
 Better to look for alternative product if your employer can't stomach one 
 Linux box in your office.
 
 Better to look for an alternative employer :)
 
 -- 
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
 
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Re: [asterisk-users] Mailing a fax with mutt does not succeed

[Andre Courchesne - Gmail]

Have you tried calling a bash script that in turns calls mutt. That way you 
could debug much easier, adding echo to a log file.

Sent from my iPhone

On 2013-06-20, at 5:27 PM, Daniel - Asterisk earohua...@gmail.com wrote:

 Hello jg:
  
 When mutt is called from Asterisk's dialplan there's no output at mail.log
  
 When I use:
 echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a 
 ${FAXDEST}/${tempfax}  /tmp/ocurrencias.txt 21
 replacing FAXDEST and TEMPFAX with proper values, the output is as follows:
  
 Jun 20 16:16:16 SERVER-NAME sendmail[21276]: My unqualified host name 
 (SERVER-NAME) unknown; sleeping for retry
 Jun 20 16:17:16 SERVER-NAME sendmail[21276]: unable to qualify my own domain 
 name (SERVER-NAME) -- using short name
 Jun 20 16:17:16 SERVER-NAME sendmail[21276]: r5KLHGgk021276: from=root, 
 size=116501, class=0, nrcpts=1, msgid=20130620211615.GA21267@SERVER-NAME, 
 relay=root@localhost
 Jun 20 16:17:17 SERVER-NAME sm-mta[21285]: r5KLHGNY021285: 
 from=root@SERVER-NAME, size=116646, class=0, nrcpts=1, 
 msgid=20130620211615.GA21267@SERVER-NAME, proto=ESMTP, daemon=MTA-v4, 
 relay=localhost [127.0.0.1]
 Jun 20 16:17:17 SERVER-NAME sendmail[21276]: r5KLHGgk021276: 
 to=earohua...@gmail.com, ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01, 
 mailer=relay, pri=146501, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent 
 (r5KLHGNY021285 Message accepted for delivery)
 Jun 20 16:17:19 SERVER-NAME sm-mta[21287]: STARTTLS=client, 
 relay=gmail-smtp-in.l.google.com., version=TLSv1/SSLv3, verify=FAIL, 
 cipher=RC4-SHA, bits=128/128
 Jun 20 16:17:20 SERVER-NAME sm-mta[21287]: r5KLHGNY021285: 
 to=earohua...@gmail.com, ctladdr=root@SERVER-NAME (0/0), delay=00:00:03, 
 xdelay=00:00:03, mailer=esmtp, pri=236646, relay=gmail-smtp-in.l.google.com. 
 [173.194.76.26], dsn=2.0.0, stat=Sent (OK 1371763040 f6si834075qaf.111 - 
 gsmtp)
 ocurrencias.txt is empty also.
  
 Elder Arohuanca
  
 
 
 On Wed, Jun 19, 2013 at 3:12 PM, jg webaccou...@jgoettgens.de wrote:
 More things to try:
 
 (1) Is there any entry in /var/log/maillog (or equivalent log file)? If so, 
 mutt basically works and the messages should give some clues.
 (2) What happens if you call mutt without any attachments?
 
 I am using mutt in exactly the same way and it works.
 
 jg
 
 Am 19.06.2013 21:50, schrieb Daniel - Asterisk:
 Hi Andre:
  
 I added echo to provide STDIN, I'm sure on variable contents, please see 
 bellow
  
  
 Hello Steve,
  
 1. I've just addd echo at my sentence, please see output bellow.
 2. Asterisk is executing as root, I think Asterisk has access to read TIF 
 files since I've used ls, chmod, cp  mv from Asterisk's CLI with '!' 
 character.
 3. I don't get you, please give some advice to try using Verbose instead 
 System
 4. I don't know how to get this, but I'm using /usr/bin/mutt as you can see 
 bellow.
 5. I have redirected output of System this way : System(echo | 
 /usr/bin/mutt -s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}  
 /tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
  
  
 DIALPLAN:
 [ Context 'default' created by 'pbx_config' ]
   '*95' =  1. NoOp(trying to send a fax to an email) 
 2. Set(FAXDEST=/tmp/faxes)
 3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11}) 
 4. NoOp(file name is: ${tempfax}) 
 5. Goto(incoming-fax,fax,7)  
  
 [ Context 'incoming-fax' created by 'pbx_config' ]
   'fax' =  1. Verbose(3,Incoming fax)
   ...
 5. ReceiveFax(${FAXDEST}/${tempfax})  
 6. Verbose(3,- Fax receipt completed with status: 
 ${FAXSTATUS}) 
 7. System(echo | /usr/bin/mutt -s New fax 
 earohua...@gmail.com -a ${FAXDEST}/${tempfax}) 
 8. NoOp(System command status is: ${SYSTEMSTATUS}) 
 9. Hangup()
  
 ASTERISK CLI OUTPUT:
 -- Goto (default,*95,1)
 -- Executing [*95@default:1] NoOp(SIP/40106-1ea1, trying to send 
 a fax to an email) in new stack
 -- Executing [*95@default:2] Set(SIP/40106-1ea1, 
 FAXDEST=/tmp/faxes) in new stack
 -- Executing [*95@default:3] Set(SIP/40106-1ea1, 
 tempfax=20130619.tif) in new stack
 -- Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name is: 
 20130619.tif) in new stack
 -- Executing [*95@default:5] Goto(SIP/40106-1ea1, 
 incoming-fax,fax,7) in new stack
 -- Goto (incoming-fax,fax,7)
 -- Executing [fax@incoming-fax:7] System(SIP/40106-1ea1, echo | 
 /usr/bin/mutt -s New fax earohua...@gmail.com   -a 
 /tmp/faxes/20130619.tif) in new stack
 -- Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System 
 command status is: APPERROR) in new stack
 -- Executing [fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in 
 new stack
  
  
 Elder D. Arohuanca
 Lima - Peru
 
 
 On Wed, Jun 19

Re: [asterisk-users] Fw: Stress testing Asterisk

[Robert-GMAIL]

I believe there are options for rtp / audio..

Look at pcap play and rtp echo...

Transcoding would be another beast - if you are allowing it

Sent from my iPhone 5

On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote:

 From the little experience I have I do not think that that is a good way of 
 testing the quality of voice. SIP only initiates and eventually terminates 
 the call, once that the call is connected, SIP and therefore Asterisk are no 
 longer involved. Once the call is connected it is assigned to a trapsport 
 layer protocol such as RTP. RTP is the actual protocol that delivers the 
 voice call between endpoints. I  believe that the setup of your network, QoS, 
 codecs etc... determine the voice quality of your system.
 
  
 - Forwarded Message -
 From: Mitul Limbani mi...@enterux.in
 To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - 
 Non-Commercial Discussion asterisk-users@lists.digium.com 
 Sent: Wednesday, May 22, 2013 3:23 PM
 Subject: Re: [asterisk-users] Stress testing Asterisk
 
 I have a question here.
 
 How can we test the quality of voice upon increasing the call load?
 
 Can we try passing a voice file using sipp and record the same in dial plan 
 record application ? Is this reliable enough to simulate near real world 
 scenario?
 
 Mitul
 
 On Wednesday, May 22, 2013, Tommy Cooper wrote:
 Thank you for your help I finally solved this issue. Is it possible that my 
 setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core 
 using 3.5 GHz, and 1Gb of RAM?
 
 - Forwarded Message -
 From: Marie Fischer ma...@vtl.ee
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com 
 Sent: Wednesday, May 22, 2013 1:16 PM
 Subject: Re: [asterisk-users] Stress testing Asterisk
 
 
 On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:
 
  Hi,
  I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is 
  generating are failing. I am trying to run Sipp on the same machine as 
  Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
 
 Do you have a peer and extension configured for SIPP in your Asterisk 
 configuration? You also needat least the -s extension_to_dial option on 
 your sipp command line.
 http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has
  some simple instructions which should get you started.
 If the calls still fail, Asterisk console output would be helpful.
 
 
 
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 -- 
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel, 
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422
 
 
 
 
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[asterisk-users] any video applications available

[Jimmy Chang(Gmail)]

We found this URL: http://sourceforge.net/projects/asteriskvideo/
But these applications seem too old for Asterisk 11.

Are there any video applications for Asterisk 11?
We need these applications to implement IVVR.

Or any other solution is to be appreciated.

Thanks in advance.

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Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif

[Robert-GMAIL]

Might also want to check the google hasnt detected an unusual login and is 
asking for the ip to be accepted.

Log in to gmail with that account and check

Sent from my iPhone 5

On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote:

 Josue Freitas wrote:
 Thank you!
 
 What about the XMPP traffic? Even when I place calls using GV there's no
 XMPP traffic on 5222.
 
 Do I really need to have the XMPP port (5222) open in the firewall?
 
 Asterisk acts as an XMPP client. It establishes an outgoing connection to 
 port 5222 of the Google Talk XMPP server. No incoming connections occur.
 
 -- 
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org
 
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[asterisk-users] Function DB_KEYS()

[Al Efron [gmail]]

Hi All,

Anyone know how to use the function DB_KEYS()?

Info on this is non-existant on the net incl. the wiki and there are
absolutely NO examples of it anywhere. I was hoping that unlike the other
DB functions, this is able to get the Key for a given Value OR at least
list ALL keys of a given Family Tree through which we can maybe iterate and
get the values of each key etc.

Speaking of which, it WOULD be quite cool if there was a function that
could do as above, i.e. find the key(s) if instead of a value lookup for a
given key, a key was returned for a given value or pattern of a known
value

Thx
\a
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Re: [asterisk-users] Function DB_KEYS()

[Al Efron [gmail]]

Ok, nevermind. Got it! Does at least one of the things I needed. Now would
be great to have a function that does the opposite ...and yes, I do know
about func_odbc, my current need just isn't big enough to setup a local
MySQL/PostGreSQL etcwas hoping to get this out of the built in DB. I
guess the next step is to maybe use AGI


On Mon, Jan 21, 2013 at 5:10 PM, Al Efron [gmail] all.efor...@gmail.comwrote:

 Hi All,

 Anyone know how to use the function DB_KEYS()?

 Info on this is non-existant on the net incl. the wiki and there are
 absolutely NO examples of it anywhere. I was hoping that unlike the other
 DB functions, this is able to get the Key for a given Value OR at least
 list ALL keys of a given Family Tree through which we can maybe iterate and
 get the values of each key etc.

 Speaking of which, it WOULD be quite cool if there was a function that
 could do as above, i.e. find the key(s) if instead of a value lookup for a
 given key, a key was returned for a given value or pattern of a known
 value

 Thx
 \a

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Re: [asterisk-users] Detect Low Quality Calls - Realtime

[Robert-GMAIL]

Sometimes just the act of collecting performance data degrades the quality

Sent from my iPhone 5

On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote:

 Thanks
 
 What would you use to measure jitter / packetloss in real time?
 
 
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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Robert-GMAIL]

Good luck! Finding the right person at VZ has always been a beef of mine


Sent from my iPhone 5

On Jan 5, 2013, at 11:12 AM, Logan Bibby lo...@keobi.com wrote:

 Does anyone have a good contact for their sales? I've attempted calling their 
 Enterprise sales a few times and was just spun around in circles. Having a 
 sales rep I can just call would be awesome.
 
 - Logan
 
 
 On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young myo...@acsacc.com wrote:
 - Original Message -
  From: Matthew J. Roth mr...@imminc.com
 
  At least Verizon maintains a consistent customer experience.  ; )
 
  Overall, we've found the service to be reliable and stable, but when
  there are problems or changes needed you're dealing with Verizon and
  the
  w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y.
 
 Haha... that is funny... it is sooo true.
 
 Well, you are right.  Once it is working, it is usually pretty stable.  Just 
 a pain in the butt when things are not working.  Hopefully we can get 
 through the Field Trial and that is all I have to worry about for a while.
 
 Thanks Matthew for all the encouragement as I go down this temporary (I 
 hope) unpleasant path.
 
 Michael
 
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 -- 
 Best regards,
 Logan
 
 Logan Bibby, CEO
 Keobi Communications
 Tuscaloosa, Alabama
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Re: [asterisk-users] Detect Low Quality Calls - Realtime

[Robert-GMAIL]

Asterisk sip show peers lists the qualify value in ms (milliseconds).

Please read up on this and the setting for it in sip.conf config file

Sent from my iPhone 5

On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote:

 Joachim, thanks for the reply
 - delay you can somewhat estimate prior to the call (with qualify for example)
 Pls be explicit. How do I use qualify to measure delay
 
 -  The jitter / packetloss you can only figure out when the call is already 
 up for a while. 
 what would you use to measure jitter / packetloss in real time?
 
 
 
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Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription

[A E [Gmail]]

I suspected as much :)

Well, it IS a calling card; people call an access number, dial an
international number.

Assuming typical ALOC to be 8 mins which is seen quite often in
International calls esp. in ethnic communities, and since the service
hasn't launched yet, it's hard to tell what the incoming traffic will be
like but in order for us to purchase the channel packs, we do need to
figure out the ratio of over-subscription we can use for the number of
channels to buy so while I understand it's a little vague, just wanted to
hear from people who're running similar services and what is their actual
channel usage and if they have consciously designed it using an assumption
for this ratio or they just buy more channels and/or DIDs looking at
historical data (or customer complaints)



On Sat, May 26, 2012 at 8:46 PM, Don Kelly d...@donkelly.biz wrote:

 I don’t think it’s possible to suggest a ratio without knowing what your
 actual application “similar to calling card services” is.

 --Don

 Don Kelly

 PCF Corp
 People Come First
 651 842-1000
 651 842-1001 fax

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
 *Sent:* Saturday, May 26, 2012 5:13 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH
 Users Help
 *Subject:* [asterisk-users] Common/Reasonable Assumption on DID/Channel
 over-subscription

 ** **

 Hello All,

 ** **

 just throwing this out there. What are people generally using these days
 when designing their services, esp. those that require a user to call a DID
 to access their system, similar to calling card services. There was a time
 when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
 channels bought in SMB with IP-PBX. 

 ** **

 I believe this would have changed today and assuming a service is pretty
 popular, the ALOCs are longer due to cheaper rates and convenience of
 calling. Does anyone have any real world numbers they can share? Is 10 to 1
 a good ratio to ensure a user practically never gets a circuits are busy?
 

 ** **

 Thanks in advance

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[asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription

[A E [Gmail]]

Hello All,

just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX.

I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have any real world numbers they can share? Is 10 to 1
a good ratio to ensure a user practically never gets a circuits are busy?

Thanks in advance
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Re: [asterisk-users] Rate sheet normalization

[A E [Gmail]]

On Mon, Mar 12, 2012 at 6:52 PM, Markus unive...@truemetal.org wrote:

 Hi,

 this question is not Asterisk specific, but since there are so many
 experts present on this list, maybe its OK to ask anyways.

 I'm having a hard time normalizing rate sheets from different providers.
 What I mean with this: the goal is to always get the cheapest rate for a
 given destination. What I would like to do is throw like 10 rate sheets
 from different providers together and as output get a single rate sheet
 with only the cheapest rates. However, some providers are listing a
 country, lets say Germany, as code 49 with a specific rate, and another
 provider will list each city individually, and each code separately, e.g.
 Berlin 4930, Hamburg 4940 etc., and probably different cities have
 different rates as well. Now, if the 49 route of the first provider is
 cheaper, my system (a2billing) will still use the more expensive 4930
 code because it is more specific.

 I'm looking for some awesome, smart tool that will automatically
 normalize all these code differences and output a clean ratesheet with
 only the cheapest rates.

 Does such a thing exist? I wonder how everyone else is normalizing their
 different rate sheets. With a homebrewn script?

 Thanks!


Markus,

you're not the first person and certainly not the last person who's ever
asked about this. I had tried this on several mailing lists a little while
ago.  A tool that could handle 10 or maybe even 5 provider rate-sheets all
of which can potentially completely differ in formats from each other. Even
worse are the rate update sheets from each provider which are many a times
different from the initial rate sheets that the provider may have given you
and then again they will differ from the rate updates from the remaining 4
providers you've just painstakingly inserted into your DB.

Given the popularity of Asterisk and other popular OSS based telephony
platforms with several successful businesses running 100s of millions of
minutes, you'd think at least a few have sorted this problem out. But I
believe those who have, never respond to these emails as it took them quite
a bit of effort to create such a tool and aren't willing to just give it
away.

Just what I have observed (and was even blatantly told by someone on some
mailing list, can't remember exactly)

You may have to advertise in the commercial / business list or offer a
bounty. There are several commercial solutions available but I think they
all come as a feature of a larger billing/rating/routing platform
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Re: [asterisk-users] Gtalk/Jabber Issue

[A E [Gmail]]

On Mon, Feb 21, 2011 at 1:21 AM, Vladimir Mikhelson v...@mikhelson.comwrote:

 William,

 It still looks like something is not properly set with your account on
 Google Voice.  Have you had a chance to follow the recommendations I
 gave you earlier in the thread?

 If the account is properly set the dial string will need to look like
 this,  gtalk/jabber-conf-section-name/+$OUTNUM$@voice.google.com
 where $OUTNUM$ is a called number in the international format.

 On the receiving end the call will come with an empty CID Number, but
 with the CID Name which looks like this:
 +1551...@voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM=

 Just cut all prior to @ as a CID Number. See
 https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google

 Also you do not need to wait 5 seconds. 1 or 2 is sufficient.

 -Vladimir


 This is a really old thread but I am having the same issues as William was
having. The incoming call just doesn't hit the context in extensions.conf. I
see the call come in on jabber...but I've tried almost 4-5 different
variations of handling the call in extensions.conf from examples on the web,
but nothing happens. I'm on 1.8.5.0.

BTW, there is no Google Voice involved. and I'm calling from from a gmail
based gtalk client. Also, I can successfully make an outbound call. Just the
inbound isn't working :( Any help please?

Currently my incoming dial-plan is:
[gtalk-in]
exten = s,1,Answer()
same = n,Wait(2)
same = n,SendDTMF(1)
same = n,Dial(SIP/2000,20)

and I have tried a whole bunch of stuff in jabber.conf and gtalk.conf but
nothing seems to cut it. I have also tried using matching my email address
(called gtalk a/c) to match in the exten as opposed to 's' extension and
that doesn't work either.

gtalk.conf
--
[general]
context=gtalk-in
bindaddr=0.0.0.0
externip=my external address
allowguest=yes

[guest]
disallow=all
allow=ulaw
context=gtalk-in
connection=asterisk

[aeg74]
username=ae...@gmail.com
disallow=all
allow=ulaw
context=gtalk-in
connection=asterisk

jabber.conf

[general]
debug=yes
autoprune=yes
autoregister=yes

[asterisk]
type=client
serverhost=talk.google.com
username=all.efor...@gmail.com/Talk
secret=my secret
port=5222   ; Port to use defaults to 5222
usetls=yes  ; Use tls or not
usesasl=yes ; Use sasl or not
buddy=ae...@gmail.com
status=available
statusmessage=On Asterisk
timeout=100

*This is the debug on jabber*
JABBER: asterisk INCOMING: iq type=set to=
all.efor...@gmail.com/Talk17BFE21F id=CA051C15DD949454 from=
ae...@gmail.com/gmail.320B5151jin:jingle action=session-initiate
sid=c1901211999 initiator=ae...@gmail.com/gmail.320B5151
xmlns:jin=urn:xmpp:jingle:1jin:content name=audio
creator=initiatorrtp:description media=audio
xmlns:rtp=urn:xmpp:jingle:apps:rtp:1rtp:payload-type id=103
name=ISAC clockrate=16000rtp:parameter name=bitrate
value=32000//rtp:payload-typertp:payload-type id=104 name=ISAC
clockrate=32000rtp:parameter name=bitrate
value=56000//rtp:payload-typertp:payload-type id=119 name=ISACLC
clockrate=16000rtp:parameter name=bitrate
value=4//rtp:payload-typertp:payload-type id=99 name=speex
clockrate=16000rtp:parameter name=bitrate
value=22000//rtp:payload-typertp:payload-type id=97 name=IPCMWB
clockrate=16000rtp:parameter name=bitrate
value=8//rtp:payload-typertp:payload-type id=9 name=G722
[Jul 18 23:36:15]
JABBER: asterisk INCOMING: clockrate=16000rtp:parameter name=bitrate
value=64000//rtp:payload-typertp:payload-type id=102 name=iLBC
clockrate=8000rtp:parameter name=bitrate
value=13300//rtp:payload-typertp:payload-type id=98 name=speex
clockrate=8000rtp:parameter name=bitrate
value=11000//rtp:payload-typertp:payload-type id=3 name=GSM
clockrate=8000rtp:parameter name=bitrate
value=13200//rtp:payload-typertp:payload-type id=100 name=EG711U
clockrate=8000rtp:parameter name=bitrate
value=64000//rtp:payload-typertp:payload-type id=101 name=EG711A
clockrate=8000rtp:parameter name=bitrate
value=64000//rtp:payload-typertp:payload-type id=0 name=PCMU
clockrate=8000rtp:parameter name=bitrate
value=64000//rtp:payload-typertp:payload-type id=8 name=PCMA
clockrate=8000rtp:parameter name=bitrate
value=64000//rtp:payload-typertp:payload-type id=117 name=red
clockrate=8000/rtp:payload-type id=106 name=
[Jul 18 23:36:15]
JABBER: asterisk INCOMING: telephone-event
clockrate=8000//rtp:descriptionp:transport xmlns:p=
http://www.google.com/transport/p2p//jin:content/jin:jingleses:session
type=initiate id=c1901211999 initiator=ae...@gmail.com/gmail.320B5151
xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=
http://www.google.com/session/phone;pho:payload-type id=103 name=ISAC
bitrate=32000 clockrate=16000/pho:payload-type id=104 name=ISAC
bitrate=56000 clockrate=32000/pho:payload-type id=119 name=ISACLC
bitrate=4 clockrate=16000/pho:payload-type id=99 name=speex
bitrate=22000 clockrate=16000/pho:payload-type id=97

Re: [asterisk-users] Google Voice receiving call problem

[A E [Gmail]]

On Thu, Jun 23, 2011 at 7:58 AM, Tim Panton t...@westhawk.co.uk wrote:


 On 15 Jun 2011, at 23:29, Kevin P. Fleming wrote:

  On 06/15/2011 04:40 PM, Elliot Murdock wrote:
  Hello,
 
  Yes, the issue I am having is currently only with Google Talk.  Wonder
  if what development will be made to fix this issue.
 
  At some point it will be fixed, and then Google will break it again.
 Google Talk/Google Voice connections to Asterisk will always be at the mercy
 of Google changing the protocol, which they do whenever they feel like it
 and with no warning. In other words, you better not be relying on it for
 critical communications, and you'll need to be patient when it breaks...
 because the developers can't just drop everything and fix it when Google
 changes the protocol.
 
  --

 A quick (uneducated) look at the packet, I think google have added some
 jingle compatibility to gtalk.

 The packet invite now contains 2 nodes - one in the jingle namespace and
 one in the google/session namespace
 this confuses  asterisk and it passes the call to _neither_ .
 I'm not up on iksemel - but I think that if it were told to match on either
 node, not just the first one things might work again

 The good news is that it supports a load of nice codecs now, including g722
 :-)


 Tim.

 Tim Panton - Web/VoIP consultant and implementor
 www.westhawk.co.uk


 So I guess incoming calls from gTalk aren't working then? (using v1.8.5.0)
I am having the exact same issue as the OP where the outgoing calls work
fine but not incoming which never hit any context within Asterisk and the
calling party only continues to hear a ringback even thought I can see the
jabber debug output for the incoming call on the console.
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
  hello people,
 
  I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
 some
  reason I have noticed that only after a few test calls, the asterisk
 process
  is running between 95% - 99.9% CPU when there's absolutely nothing on the
  system. This is a clean Asterisk system in an internal network with
 nothing
  else on it with no calls on it but it's still sitting with 96% CPU.
 
  I'm not a developer so not that ept with using debug tools etc to figure
 out
  why it's doing that. Could anyone please tell me how I can figure out why
  it's doing this and/or help debug this. Makes no sense for it to be using
  CPU with nothing happening on the system

 The first thing I'd do is run 'top', press shift H, and see what is/are
 the offending thread(s).

 Is it a single thread? Two? More?

 Is it all user time? Much of it is system time?

 If you strace the PID of the top thread (strace -p PID), what do you
 see?


 Hi Tzafrir,

thanks for the comments and suggestions. So I'd done all of that and what
I'd found was

- After I'd done Shift-h, There was only one / single thread that was taking
all of the CPU
- 33% was Sser and 66% was System times
- when I'd run an strace on the PID of the offending thread it just rolled
some message past my screen which I couldn't capture and can't remember what
it said :(

Anyway I've killed that process, updated the packages the system, upgraded
to 1.8.4.4 and will give it another shot and see what happens. Would've
helped if I'd kept the system as it was so people could help me figure out
what was going on, but the fact that it stopped responding to commands which
were trying to kill the hung channels, reloading configs, or even trying to
stop the system wouldn't work is bizarre. I hope the developers pay
attention to that.
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
  On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
 wrote:
 
   On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
hello people,
   
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
   some
reason I have noticed that only after a few test calls, the asterisk
   process
is running between 95% - 99.9% CPU when there's absolutely nothing on
 the
system. This is a clean Asterisk system in an internal network with
   nothing
else on it with no calls on it but it's still sitting with 96% CPU.
   
I'm not a developer so not that ept with using debug tools etc to
 figure
   out
why it's doing that. Could anyone please tell me how I can figure out
 why
it's doing this and/or help debug this. Makes no sense for it to be
 using
CPU with nothing happening on the system
  
   The first thing I'd do is run 'top', press shift H, and see what is/are
   the offending thread(s).
  
   Is it a single thread? Two? More?
  
   Is it all user time? Much of it is system time?
  
   If you strace the PID of the top thread (strace -p PID), what do you
   see?
  
  
   Hi Tzafrir,
 
  thanks for the comments and suggestions. So I'd done all of that and what
  I'd found was
 
  - After I'd done Shift-h, There was only one / single thread that was
 taking
  all of the CPU
  - 33% was Sser and 66% was System times
  - when I'd run an strace on the PID of the offending thread it just
 rolled
  some message past my screen which I couldn't capture and can't remember
 what
  it said :(

 Just press ctrl-c .

 haha I did that but since that I did a 100 other things in my ssh window
which is only buffered for 5000 lines and those messages have gone past.


 
  Anyway I've killed that process, updated the packages the system,
 upgraded
  to 1.8.4.4 and will give it another shot and see what happens. Would've
  helped if I'd kept the system as it was so people could help me figure
 out
  what was going on, but the fact that it stopped responding to commands
 which
  were trying to kill the hung channels, reloading configs, or even trying
 to
  stop the system wouldn't work is bizarre. I hope the developers pay
  attention to that.

 Developers need some data to work with :-(

 Haha of course. Although I have a feeling it'll happen again as this is the
2nd time this has happened. Will keep the system in that state till we can
try and resolve this and capture enough info. if I had better memory, I'd
have actually remembered what the message was, but anyway, what I was trying
to say was that it's much more than just taking up all the CPU tells me
that some thread has just gone loco. But the fact the CLI and AMI commands
become unresponsive when trying to kill these zombie channels or trying to
do a core reload or core stop now etc. tells me that this is a bigger
issue than just some thread gone nuts and the channels being hung
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 7:50 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote:
  On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
 wrote:
 
   On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:

- when I'd run an strace on the PID of the offending thread it just
 rolled
some message past my screen which I couldn't capture and can't
 remember
what it said :(
  
   Just press ctrl-c .
  
  haha I did that but since that I did a 100 other things in my ssh window
  which is only buffered for 5000 lines and those messages have gone past.

 If the process / thread is in a loop, the messages tend to repeat
 themselves.

 Also: anything interesting in /var/log/asterisk/messages ?

 Yup, it surely was in some funky loop...and I wouldn't be surprised if it
was looping to check if the channels were hungup or not and ended up taking
up the entire CPUI should've tried to just kill that thread with its PID
and seen if the operation returns to normal.

No, unfortunately nothing interesting found in the logs, other than the
indication that when I tried to reload using core reload it was actually
loading the configs even though it didn't show anything on the CLI.
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[asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

hello people,

I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
reason I have noticed that only after a few test calls, the asterisk process
is running between 95% - 99.9% CPU when there's absolutely nothing on the
system. This is a clean Asterisk system in an internal network with nothing
else on it with no calls on it but it's still sitting with 96% CPU.

I'm not a developer so not that ept with using debug tools etc to figure out
why it's doing that. Could anyone please tell me how I can figure out why
it's doing this and/or help debug this. Makes no sense for it to be using
CPU with nothing happening on the system

Thanks
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk earohua...@gmail.comwrote:

 On the CLI write: sip show channels

 If there are lots of bye channels you have the same problem than me.
 I've tried waiting with the call generator -sipp- and channels
 finished when there are a few. But they're not ending faster enough
 when I send lots of concurrent calls.

 Elder

 Hi,
thanks for the response. yeah I'd checked that before and I only have 2
dialogs which seem to be part of the same call that are just sitting there
and I can't seem to get them to hang up by typing channel request hangup
all . I even tried sending a Hangup by connecting on the AMI but that
doesn't seem to be doing anything either. So this channel is sitting there
in the 'BYE' state.
Is there anyway of clearing them without having to reload/restart Asterisk?
I want to see if that's the cause of the CPU usage and I'll lose that if I
restart Asterisk.
Thanks



 2011/7/5, A E [Gmail] all.efor...@gmail.com:
  hello people,
 
  I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
 some
  reason I have noticed that only after a few test calls, the asterisk
 process
  is running between 95% - 99.9% CPU when there's absolutely nothing on the
  system. This is a clean Asterisk system in an internal network with
 nothing
  else on it with no calls on it but it's still sitting with 96% CPU.
 
  I'm not a developer so not that ept with using debug tools etc to figure
 out
  why it's doing that. Could anyone please tell me how I can figure out why
  it's doing this and/or help debug this. Makes no sense for it to be using
  CPU with nothing happening on the system
 
  Thanks
 

 --
 Enviado desde mi dispositivo móvil

 --
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Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system

[A E [Gmail]]

On Wed, Jul 6, 2011 at 1:49 AM, Faisal Hanif fai...@vopium.com wrote:

 You have to provide channel ID to command like “channel request hangup
 SIP/12316156-sad4d46a5”.

 **


Thanks, but all is also a valid keyword according to the documentation. I
think there are some bugs associated with hung channels. Nothing seems to
work when a channel is hung in that state. hanging up is not working, nor
the AMI is working in providing status etc. and when I'm on the CLI, even
core stop now doesn't work and it hands the CLI.

Something is majorly wrong. I'm going to upgrade the version to 1.8.4.4 and
see what happens


  **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
 *Sent:* Wednesday, July 06, 2011 9:50 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+
 CPU with No calls on the system

 ** **

 ** **

 On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk earohua...@gmail.com
 wrote:

 On the CLI write: sip show channels

 If there are lots of bye channels you have the same problem than me.
 I've tried waiting with the call generator -sipp- and channels
 finished when there are a few. But they're not ending faster enough
 when I send lots of concurrent calls.

 Elder

 Hi,

 thanks for the response. yeah I'd checked that before and I only have 2
 dialogs which seem to be part of the same call that are just sitting there
 and I can't seem to get them to hang up by typing channel request hangup
 all . I even tried sending a Hangup by connecting on the AMI but that
 doesn't seem to be doing anything either. So this channel is sitting there
 in the 'BYE' state. 

 Is there anyway of clearing them without having to reload/restart Asterisk?
 I want to see if that's the cause of the CPU usage and I'll lose that if I
 restart Asterisk.

 Thanks

 ** **

  

 2011/7/5, A E [Gmail] all.efor...@gmail.com:

  hello people,
 
  I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
 some
  reason I have noticed that only after a few test calls, the asterisk
 process
  is running between 95% - 99.9% CPU when there's absolutely nothing on the
  system. This is a clean Asterisk system in an internal network with
 nothing
  else on it with no calls on it but it's still sitting with 96% CPU.
 
  I'm not a developer so not that ept with using debug tools etc to figure
 out
  why it's doing that. Could anyone please tell me how I can figure out why
  it's doing this and/or help debug this. Makes no sense for it to be using
  CPU with nothing happening on the system
 
  Thanks
 

 --
 Enviado desde mi dispositivo móvil

 --
 _
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   http://lists.digium.com/mailman/listinfo/asterisk-users

 ** **

 --
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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 12:12 AM, A E [Gmail] all.efor...@gmail.com wrote:

 Hello,
 using 1.8.4. using a very simple local AGI script in bash which has only
 one line in it:

 echo -e 'STREAM FILE welcome 123 \n'

 dialplan:
 exten = 5150,1,Answer()
   same = n,Set(CHANNEL(language)=en_AU)
   same = n,AGI(testagi.sh)
   same = n,Hangup

 console output:
 -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new
 stack
 -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024,
 CHANNEL(language)=en_AU) in new stack
 -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh)
 in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
 -- Playing 'welcome' (escape_digits=1) (sample_offset 0)
 -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0
 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new
 stack
   == Spawn extension (AllPhones, 5150, 4) exited non-zero on
 'SIP/PBX-0024'

 But nothing happens...as in even when it says that it's playing the file
 (as verified in the asterisk 'full' log), I hear nothing on the phone

 What gives? spent 2 hrs Googling but nothing! :(

 Thx
 \A

  Bizarre, I found a bunch of other agi scripts in the default
directory...modified the agi-test.agi (perl script) so it played my file, no
joy! then I used a php script I found somewhere else asa tutorial to writing
AGI scripts in php, modified that to play my script and it works. I don't
get it. esp. when everything (with agi debug set on) looks exactly the same
with my bash script and this php script except that with the php script, I
see this ONE line that's extra


SIP/PBX-002bAGI Rx  STREAM FILE welcome #
-- Playing 'welcome' (escape_digits=#) (sample_offset 0)
SIP/PBX-002bAGI Tx  200 result=35 endpos=87200

that I don't see with my bash script which does this


SIP/PBX-002eAGI Rx  STREAM FILE welcome #

-- Playing 'welcome' (escape_digits=#) (sample_offset 0)

-- SIP/PBX-002eAGI Script streamcontact.sh completed, returning 0

 -- Executing [5150@AllPhones:5] Hangup(SIP/PBX-002e, ) in new
stack

So confused!!
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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:06 AM, mahesh katta maheshka...@flexydial.comwrote:



 On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] all.efor...@gmail.com wrote:

 Hello,
 using 1.8.4. using a very simple local AGI script in bash which has only
 one line in it:

 echo -e 'STREAM FILE welcome 123 \n'

 dialplan:
 exten = 5150,1,Answer()
   same = n,Set(CHANNEL(language)=en_AU)
   same = n,AGI(testagi.sh)

 same = n,Hangup

 Try this below dilaplan

 exten = 5150,1,Answer()
   same = n,Set(CHANNEL(language)=en_AU)
   same = n,AGI(testagi)
 same = n,Hangup


No deal. Doesn't find the AGI script



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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Mon, 6 Jun 2011, A E [Gmail] wrote:

  Hello,using 1.8.4. using a very simple local AGI script in bash which has
 only one line in it:

 echo -e 'STREAM FILE welcome 123 \n'

 What gives? spent 2 hrs Googling but nothing! :(


 Maybe 1.5 hrs should have been spent reading :)


touche ;)


 One line does not an AGI make.

 Did you just pull a 'Yoda' Steve?


 AGI is an interface. It consists of reading the AGI environment from STDIN
 and then, writing requests on STDOUT and reading the response from STDIN.

 Right! I did read that, the problem is how do I do this in bash?? I tried
read the result in and just post a Noop kind of a thing just to tell that I
read something, but it didn't help. I also explicitly did that in the perl
script, but doesn't work. It only works in PHP.


 If you don't follow these 3 steps in order (steps 2 and 3 can be repeated)
 then your program has violated the protocol and will not function reliably
 if at all.

 Please use an existing AGI library for the language of your choice. Nobody
 gets it right the first time.


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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.comwrote:

  On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.com
 wrote:


  AGI is an interface. It consists of reading the AGI environment from STDIN
 and then, writing requests on STDOUT and reading the response from STDIN.


 On Mon, 6 Jun 2011, A E [Gmail] wrote:

  Right! I did read that, the problem is how do I do this in bash?? I tried
 read the result in and just post a Noop kind of a thing just to tell that I
 read something, but it didn't help. I also explicitly did that in the perl
 script, but doesn't work. It only works in PHP.


 Bash would probably be my last choice of language to write an AGI with.

 Personally, I use C because it is my sharpest tool and because you can
 execute hundreds of AGIs written in C in the time it takes to load the Perl
 or PHP interpreter and parse your script.

 I suspect that the problems you are experiencing with Perl may have
 something to do with flushing STDOUT or reading the complete response from
 STDIN.

 I strongly suggest using an existing library for the language of your
 choice.


 Copy that. Not planning to write an AGI script in bash actually...it will
be written in C# running on a remote system. I was just doing a quick PoC to
figure out how would I use the stream file function to actually read audio
files over the network and even though I used to teach Perl 10+ yrs ago, I
don't do much scripting/coding for a long time, so the brain doesn't think
like a coder anymore. Just needed to try various tricks w.r.t to how can I
dynamically bring over audio files from another server, convert them to the
codec of my channel and then play/store them locally (cache if you will) but
wanted to learn the right way to do it with a local file first before I
tried something fancier. Guess I'll continue playing with the php script
that worked and once I figure the process out, will give it to the C# dev to
implement. Can't believe I wasted more than 2-3 hrs on this :(

BTW, I'd raised that a while ago, and got no conclusive response. How / what
is the best way to stream audio files (not MOH/Internet Radio/TV and what
not) inside a dialplan using AGI without comprising performance/adding
latency too much. no examples of shout/ICE I could find that show how to do
that simply by allowing me to run a web server remotely and use a shoutcast
module to play the audio right into the channel

ideas?
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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 10:39 AM, Tony Mountifield t...@mountifield.orgwrote:

 In article banlktikpxc_jk5xvhyx9akakltvl0v6...@mail.gmail.com,
 A E [Gmail] all.efor...@gmail.com wrote:
  Hello,
  using 1.8.4. using a very simple local AGI script in bash which has only
 one
  line in it:
 
  echo -e 'STREAM FILE welcome 123 \n'
 
  dialplan:
  exten = 5150,1,Answer()
same = n,Set(CHANNEL(language)=en_AU)
same = n,AGI(testagi.sh)
same = n,Hangup
 
  console output:
  -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in
 new
  stack
  -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024,
  CHANNEL(language)=en_AU) in new stack
  -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024,
 testagi.sh) in
  new stack
  -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
  -- Playing 'welcome' (escape_digits=1) (sample_offset 0)
  -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0
  -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in
 new
  stack
== Spawn extension (AllPhones, 5150, 4) exited non-zero on
  'SIP/PBX-0024'
 
  But nothing happens...as in even when it says that it's playing the file
 (as
  verified in the asterisk 'full' log), I hear nothing on the phone
 
  What gives? spent 2 hrs Googling but nothing! :(

 Firstly, you need to check that you can successfully play files
 outside of the AGI environment. Replace the AGI command with:

 same = n,Playback(welcome)

 Yes, had done that as first order of business before even trying AGI


 If that doesn't work, the problem is nothing to do with AGI.

 However, I think what else is happening is that your AGI script is
 sending the STREAM FILE command and then immediately exiting. This goes
 back to the dialplan and executes a hangup when only a tiny fraction of
 the welcome file has been played. You could test this theory in two
 different ways, as I'm not sure whether it's the exiting of the AGI or
 the subsequent hangup that is aborting the playback.


I thought so too, didn't know what to do about it


 a) Put a sleep 5 in your agitest.sh after the echo. As others have
 said, you should really use a proper library that reads responses to
 AGI commands, but for testing, a sleep will keep the AGI script alive
 while the message plays.

 This works!


 b) Put a same = n,Wait(5) after the AGI command. If the AGI leaves
 the message playing, this would give it some time to play before you
 hang up the line.

 This doesn't work

Thanks for the help. I just need some sort of a wait loop there (as I don't
really know how to pick up the 200 result=0) till the prompt finishes.
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[asterisk-users] Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?)

[A E [Gmail]]

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.com
 wrote:

 I strongly suggest using an existing library for the language of your
 choice.


 On Mon, 6 Jun 2011, A E [Gmail] wrote:

  Copy that. Not planning to write an AGI script in bash actually...it will
 be written in C# running on a remote system.

 How / what is the best way to stream audio files (not MOH/Internet
 Radio/TV and what not) inside a dialplan using AGI without comprising
 performance/adding latency too much.


 Well, C# means you're getting your data from a Windows host, so I'd fix
 that first :)

 now now. It works pretty well actually, can implement extremely complicated
logic, multi-threaded, can run as a service, and integrates with the web-app
which is all in asp.net etc. anyway, moving on


 Without knowing all the details, the options I see are:

 ) Transfer the file using HTTP, FTP, SCP, etc. You'll have to wait until
 the entire file is transferred before you can start playing.

 ) 'Stream' the file using a shared file system like NFS or Samba. If the
 'source' and 'target' hosts are on different continents this may not be
 practical. If they are in the same rack...

 ) Stream the file using a custom application. app_playback.c is only about
 550 lines (1.8.0) which includes all the standard application 'boilerplate'
 for help, cli interface, loading, unloading, etc. as well as all of
 playback's little buddies like SayAlpha, SayDigits, SayNumber, etc. so a
 custom application cribbed from app_playback.c should only be 100 lines or
 so.


Right. Had thought about all of those, but looking for something along the
lines of an application that can be invoked from inside the AGI socket
connection i.e. picking a file over the network from a fast/lite http server
(ala lighthttpd/nginx) and streaming it into the channel. So kind of like
a 'Playback/Background over the network' kind of an app so one doesn't have
to worry about bringing the file over, using NFS/SAMBA fileshares, caching
and thus avoiding excessive file i/o. Does the MP3Player application do
that? We could do that but ideally I'd like to avoid any transcoding etc. so
we can create and save files in a ulaw/g729 etc formats and then just stream
them avoiding all latency, file i/o, CPU issues.

You're right, playback/background could be modified, unfortunately I'm not a
C developer, so I might not be able to do it. But if someone knows of
something that does the above from inside an AGI connection, that'd be
awesome.

Thanks so much
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[asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

Hello,
using 1.8.4. using a very simple local AGI script in bash which has only one
line in it:

echo -e 'STREAM FILE welcome 123 \n'

dialplan:
exten = 5150,1,Answer()
  same = n,Set(CHANNEL(language)=en_AU)
  same = n,AGI(testagi.sh)
  same = n,Hangup

console output:
-- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new
stack
-- Executing [5150@AllPhones:2] Set(SIP/PBX-0024,
CHANNEL(language)=en_AU) in new stack
-- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
-- Playing 'welcome' (escape_digits=1) (sample_offset 0)
-- SIP/PBX-0024AGI Script testagi.sh completed, returning 0
-- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new
stack
  == Spawn extension (AllPhones, 5150, 4) exited non-zero on
'SIP/PBX-0024'

But nothing happens...as in even when it says that it's playing the file (as
verified in the asterisk 'full' log), I hear nothing on the phone

What gives? spent 2 hrs Googling but nothing! :(

Thx
\A
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Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[A E [Gmail]]

On Thu, May 19, 2011 at 3:19 AM, GNUbie gnu...@gmail.com wrote:

 Anyone? Please advice. Thank you.

 That's WAYY too much info for me to go through right now, and I don't know
anything about TLS registration but what I would ask for is if you have the
following lines in your sip.conf

domain=IP/FQDN of your asterisk server:TLS port

so in your case add the lines

domain=pbx.domain.com:5061

and then do a sip reload

So far, all problems I've had, have been solved because of this. At the end
of your sip.conf add those lines and it should fix your problem.

HTH
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Re: [asterisk-users] v1.8.4: Extension Not found in Context?

[A E [Gmail]]

On Wed, May 18, 2011 at 9:39 PM, A E [Gmail] all.efor...@gmail.com wrote:

 On Wed, May 18, 2011 at 9:29 PM, Paul Belanger pabelan...@digium.comwrote:

 On 11-05-18 08:01 PM, A E [Gmail] wrote:

 boxb*CLI  dialplan show Test
 [ Context 'Test' created by 'pbx_config' ]
   '' =  1. Answer()
 [pbx_config]
 2. Wait(2)
  [pbx_config]
 3. Hangup()
 [pbx_config]

 -= 1 extension (3 priorities) in 1 context. =-

 But when the call comes into boxb from box a, on box b CLI I see the msg:

 NOTICE[1613]: chan_sip.c:21581 handle_request_invite: Call from 'boxA' to
 extension '' rejected because extension not found in context 'Test'.

 WHY??

 Thanks :(

  Does the peer using 'boxA' dialplan include context 'Test'?

 You mean in its definition/declaration in sip.conf? yes. sip.conf in Box B
 looks like this:

 [boxA]
 type=peer
 host=10.0.3.5
 context=Test
 disallow=all
 allow=ulaw
 allow=g722
 allow=g729
 dtmfmode=rfc2833
 canreinvite=no
 insecure=port,invite


 Ok, this problem is fixed. Once again, it was the damn domain= line in
sip.conf

Since I was using a non-standard port i.e. 5062, just using, autodomain=yes
doesn't help. One needs to explicitly specify the local address and bindport
to be included. But the message in the console is misleading. I think I need
to open a bug/issue about this.

If I have a udpbindaddr = 10.0.3.6:5062, then autodomain keyword, should
actually be smart enough to read that and auto-include the port specified
(if specified).

Thanks
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Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so

[A E [Gmail]]

On Mon, May 16, 2011 at 10:20 PM, Shaun Ruffell sruff...@digium.com wrote:

 On Mon, May 16, 2011 at 09:26:48PM -0400, A E [Gmail] wrote:
 
  following this advice, is there a quick and minimal way to install/use
  res_timing_dahdi without having to build/compile/install the whole dahdi
  package and all the other modules associated with it? back in the zaptel
  days, I used to be able to modify the Makefile and compile JUST the
 ztdummy
  module to provide timing for meetme. Haven't touched * for a while esp.
  Zaptel/Dahdi, so not sure how it works anymore.

 In the dahdi-linux package you can edit drivers/dahdi/Kbuild and comment
 out
 every module except for dahdi.ko. So looking in that file you will see
 something like:

 obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI)   += dahdi.o
 #obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DUMMY)+= dahdi_dummy.o
 obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DYNAMIC)   += dahdi_dynamic.o
 obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DYNAMIC_LOC)   +=
 dahdi_dynamic_loc.o

 Here dahdi_dummy is commented out.  Just comment out all the other modules
 (lines that start with obj-) and leave only dahdi.o.

 dahdi.ko now automatically monitors the spans and if there isn't one
 providing
 timing, it will use the built in timing source which functions very
 similarly
 to dahdi dummy of the past.

  I'm assuming to get res_timing_dahdi, I need dahdi_dummy installed at the
  very least?

 Since dahdi-linux 2.3.0, all you need is dahdi.ko. There is no more
 dahdi_dummy module required unless you specifically install it.

  Do I need the kernel source packages like in the old days to compile
 DAHDI
  against the Kernel etc?

 You will still need the kernel sources to compile dahdi.ko against. Also
 when
 you install dahdi-tools, you will want to comment out all the lines in
 /etc/dahdi/modules so that the init script does not try to load any of the
 board drivers.


Wow! Thanks Shaun for the amazingly detailed and clear instructions. Really
appreciate it.  Let me give this a go.

Cheers :)
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Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so

[A E [Gmail]]

On Mon, May 16, 2011 at 10:27 AM, satish patel satish...@hotmail.comwrote:

  Thanks Leif,

 I had changed it to res_timing_dahdi and since last few days it seem good.

 -S

  Date: Sun, 15 May 2011 15:48:03 -0400
  From: leif.mad...@asteriskdocs.org
  To: asterisk-users@lists.digium.com

  Subject: Re: [asterisk-users] res_timing_timerfd.so Vs
 res_timing_dahdi.so
 
  On 11-05-13 11:39 AM, isr...@gmail.com wrote:
   I haven't tried with timerfd but with timer pthread 1.8 is very
 unstable
  
   I think I have seen a post to the list from kevin fleming that the same
 is for timerfd that there is a nasty bug which they haven't found the reason
 for yet
 
  My experience is that you should pretty much always use res_timing_dahdi
 unless
  you're on a platform on which you can't install DAHDI. You don't need any
  hardware to use timing from DAHDI because timing is generated by the
 kernel.
 
  My order of preference for stability is:
 
  * res_timing_dahdi
  * res_timing_timerfd
  * res_timing pthread
 
  The timerfd and pthread modules are relatively new, and sometimes people
 run
  into stability problems while using them. If you can use res_timing_dahdi
 I
  recommend you do so.
 
  Leif.
 


following this advice, is there a quick and minimal way to install/use
res_timing_dahdi without having to build/compile/install the whole dahdi
package and all the other modules associated with it? back in the zaptel
days, I used to be able to modify the Makefile and compile JUST the ztdummy
module to provide timing for meetme. Haven't touched * for a while esp.
Zaptel/Dahdi, so not sure how it works anymore. I'm assuming to get
res_timing_dahdi, I need dahdi_dummy installed at the very least? Do I need
the kernel source packages like in the old days to compile DAHDI against the
Kernel etc?

Thx so much
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[asterisk-users] Rates Importer Tool

[A E [Gmail]]

Hi All,

new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?

Thanks so much in advance
aeg
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Re: [asterisk-users] Rates Importer Tool

[A E [Gmail]]

On Mon, May 9, 2011 at 3:05 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.com wrote:

  I know most billing software sell this as a monthly service.  You get
 cd-rom every month where they have collected the published tarrif tables
 filed with the FCC. You load it on the software to analyze call costs.   I’m
 guessing this is a lot of labor hours/manual work thus they charge for
 providing it.  In particular I am thinking of InforTel for Windows.


That's interesting. Wasn't aware of such a thing...if these subscriptions
ad/or software are reasonably priced then we might still be interested in
having a look at it. What specific product of InforTel were you referring
to?
Thanks
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Re: [asterisk-users] Rates Importer Tool

[A E [Gmail]]

On Mon, May 9, 2011 at 7:58 PM, Markus unive...@truemetal.org wrote:

 Hi,

  new to the list. Wondering if anyone has / knows of, a good rate importer
  tool that can be used to standardize and normalize the ratesheets / rate
  decks etc. obtained from various carriers so they can be analysed and
  imported into a DB or be saved as a CSV or something?

 I'm using a2billing (http://www.a2billing.org), a free of charge and
 complete call shop web-based PHP application for Asterisk. Very buggy
 overall but I couldn't find anything better (which is free of charge) yet.
 Anyway, it gets the job done. I'm uploading the rate sheets to a Linux box
 respectively download them directly to the box and then use a shell script
 for each provider's rate sheet to properly order to fit into the a2billing
 format, a la:

 wget http://www.provider.com/rates/premium.csv
 cat premium.csv | grep \1\,\1\  temp.csv
 cat temp.csv | cut -d , -f 3  tempcode.csv
 cat temp.csv | cut -d , -f 1  tempdest.csv
 cat temp.csv | cut -d , -f 6  temprate.csv
 paste -d , tempcode.csv tempdest.csv temprate.csv temprate.csv | tail
 -n+2 | sed 's/^\/\00/g'  Provider.PREMIUM.$DATE.csv
 unix2dos Provider.PREMIUM.$DATE.csv
 scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
 rm temp.csv tempcode.csv tempdest.csv temprate.csv premium.csv

 This fetches and orders the rate sheet properly and uploads it to my home.
 Then I just log into a2billing and upload the rate sheet there, done with
 a few clicks. But you could also create a new ratecard directly in MySQL
 and store the rates there directly if you want to. a2b stores all rates in
 a MySQL DB. You can then choose least cost routing between different
 providers etc.

 Also, when a provider only supplies XLS instead of CSV, I use a script
 like the following, utilizing xlhtml:

 xlhtml -csv -xp:0 Provider.xls | cut -d , -f 1-3  temp.csv
 cat temp.csv | cut -d , -f 3  temp2.csv
 paste -d , temp.csv temp2.csv | tail -n+3 | sed 's/^/00/g' 
 Provider.PREMIUM.$DATE.csv
 unix2dos Provider.PREMIUM.$DATE.csv
 scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
 rm temp.csv
 rm temp2.csv

 Regards.

 Hello Markus,
thanks for sharing. I am looking into A2Billing myself at the moment. Don't
really need most of the functionality in it, but will check out its rates
import tool although I'm not sure it can handle rate updates but seems like
something to check out. Although like I'd said in my OP, this is mostly for
the business people to be able to visualize the rates and analyse them them
more than anything else and judging from the extra hacking involved in
getting these rates to be ready to be imported into A2Billing even seems too
complicated for the business people be able to do on their own, and I don't
want to have to sit and normalize it for them every time there's a rate
update. But will look more into this. Thanks again for putting up your
script and trying to help out :)
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Re: [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI

[A E [Gmail]]

On Tue, May 3, 2011 at 4:41 AM, Thorsten Göllner t...@ovm-group.com wrote:

  Am 02.05.2011 15:59, schrieb A E [Gmail]:

  On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.comwrote:

 Hello All,

  Probably a silly question, but we're wondering if people have had any
 experience and have data to demonstrate if the performance of the Asterisk
 system might suffer in terms of latency etc. if we're to have it retrieve
 sound files from a database using odbc as opposed to storing them locally on
 the filesystem. Note, these are not prompts...these are sound files that are
 being created through a web-app and being stored in the DB as BLOB or
 similar datatype that's good/efficient to store audio/video files in a DB.
 We need these be made available through the asterisk system to play over the
 phone. Although the DB uses a SAN, the Asterisk System has no connectivity
 to the SAN but is connected on the same physical ethernet switch with a
 multi-Gbps backplane.

  The way the system is being designed, it's possible for us to end up
 with 000s of these sound files stored in the DB, not to mention several
 asterisk systems in a pool/cluster/farm requesting these files, so using the
 local filesystem might not be scalable or efficient.

  Any advice/comments/suggestions welcome :)


   Just realised that this can better be described another way:

  What we're essentially trying to do is be able to do any one of these

  a) stream an audio/video file stored in the DB via AGI into the current
 channel so that it plays on the phone

  OR

  b) Do something like what Realtime Voicemail does, where it gets the file
 from the DB, saves as a temp file in the user mailbox directory and then
 plays it to the caller but this needs to happen through AGI, something along
 the lines of readsql (a la func_odbc) inside of AGI

  OR

  c) Anything else that's better than a) and b) above that someone can
 suggest.

  P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE
 which seems to be the only solution we can think of right now, other than of
 course having the DB machine exporting the SAN volume as an NFS share for
 the Asterisk server to mount, but that sounds like it'll be bad for
 performance?


 Mediafiles are stored on SAN and the DB stores references to the files in
 the SAN. I do not see a problem doing it this way. It's scaleable and
 efficient. Where do you suppose to run into problems?


Well the problem as explained above is that the * machine does NOT have
direct access to the SAN. So we cannot mount the SAN volume on the * machine
as a shared drive. It'll have to either be shared by the DB server for the *
machine to read from, OR we can use the GET SOUNDFILE / PUT SOUNDFILE AGI
AddOn to copy the sound files from the DB machine to the * machine to play
it over the phone. Just wondering if that will have any major performance
impact
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[asterisk-users] Retrieving sound files from DB as opposed to filesystem

[A E [Gmail]]

Hello All,

Probably a silly question, but we're wondering if people have had any
experience and have data to demonstrate if the performance of the Asterisk
system might suffer in terms of latency etc. if we're to have it retrieve
sound files from a database using odbc as opposed to storing them locally on
the filesystem. Note, these are not prompts...these are sound files that are
being created through a web-app and being stored in the DB as BLOB or
similar datatype that's good/efficient to store audio/video files in a DB.
We need these be made available through the asterisk system to play over the
phone. Although the DB uses a SAN, the Asterisk System has no connectivity
to the SAN but is connected on the same physical ethernet switch with a
multi-Gbps backplane.

The way the system is being designed, it's possible for us to end up with
000s of these sound files stored in the DB, not to mention several asterisk
systems in a pool/cluster/farm requesting these files, so using the local
filesystem might not be scalable or efficient.

Any advice/comments/suggestions welcome :)
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Re: [asterisk-users] HA Asterisk

[A E [Gmail]]

On Mon, May 2, 2011 at 12:07 AM, Kaushal Shriyan
kaushalshri...@gmail.comwrote:


 Hi Jim,

 Thanks for the explanation, I have couple of questions here.

 1) Does the xorcom box support *8 Port PRI E1 Interface*. ?
 2) Also the Primary and Secondary Asterisk Server can be any server which
 will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow)
 Application and customizable or do i also need to buy this from Xorcom ? Not
 sure i understand that.
 3) How does the xorcom box communicate with the Asterisk Server which do
 not contain any PRI Card inside the system.

 Much Appreciated.

 Thanks and Regards,

 Kaushal


Kaushal,

1) it's all clearly explained on their page. Looking at the video, one can
tell they have 8 PRI ports on that box and 8 FXS ports and there's space for
3 further 8-channel modules that can be added. You can get an XR0111 for 8
PRIs (or XR0015 for BRI):
http://www.xorcom.com/telephony-interfaces/astribank-models.html

2) It also states there that the Astribank's drivers have been a part of
Zaptel/DAHDI since early 2006. Which means that it's MOST likely compatible
with any home-baked Asterisk installation without the need to buy Xorcom
Servers.

3) Lastly, it clearly uses these Astribank drivers in DAHDI to make the
Astribank channel bank as an external hardware to Asterisk to talk back and
forth. Since USB is a physical connection between the two, I'm sure if a
server is down, the software in Astribank can detect the lack of
connectivity on that USB port (i.e. voltage) as well as it might realise
there's no communication between it and the Astribank driver in DAHDI on the
Asterisk server.

One should not just try and get answers the easy way. You could've figured
all this out in 5 mins just like I did...not that I'm saying I'm really
smart ;)

Anyway, hope it helps :)

Now, I wonder what're the alternatives that people have been using for
Asterisk HA other than commercially available solutions like HAAST and
Astribanks assuming that kaushal is right and SCF isn't production ready
yet. Anyone wants to chime in here with a solution built with readily
available linux software like heartbeat , linux-ha, shared filesystems,
filesystem replication and of course asterisk realtime? My requirement might
be more along the lines of having several asterisk servers in a farm/pool
without actually caring about the failover, so it might not even matter for
me to worry about all of this, but I'm still curious as to what people are
doing out there.

Cheers
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Re: [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI

[A E [Gmail]]

 Just realised that this can better be described another way:

 What we're essentially trying to do is be able to do any one of these

 a) stream an audio/video file stored in the DB via AGI into the current
 channel so that it plays on the phone

 OR

 b) Do something like what Realtime Voicemail does, where it gets the file
 from the DB, saves as a temp file in the user mailbox directory and then
 plays it to the caller but this needs to happen through AGI, something along
 the lines of readsql (a la func_odbc) inside of AGI

 OR

 c) Anything else that's better than a) and b) above that someone can
 suggest.

 P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which
 seems to be the only solution we can think of right now, other than of
 course having the DB machine exporting the SAN volume as an NFS share for
 the Asterisk server to mount, but that sounds like it'll be bad for
 performance?

 Thanks again


No takers? :(
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Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI

[A E [Gmail]]

On Mon, May 2, 2011 at 2:30 PM, Danny Nicholas da...@debsinc.com wrote:

--

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
 *Sent:* Monday, May 02, 2011 1:23 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Retrieving/Streaming audio/video files
 from DBusing over AGI





  Just realised that this can better be described another way:



 What we're essentially trying to do is be able to do any one of these



 a) stream an audio/video file stored in the DB via AGI into the current
 channel so that it plays on the phone



 OR



 b) Do something like what Realtime Voicemail does, where it gets the file
 from the DB, saves as a temp file in the user mailbox directory and then
 plays it to the caller but this needs to happen through AGI, something along
 the lines of readsql (a la func_odbc) inside of AGI



 OR



 c) Anything else that's better than a) and b) above that someone can
 suggest.



 P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which
 seems to be the only solution we can think of right now, other than of
 course having the DB machine exporting the SAN volume as an NFS share for
 the Asterisk server to mount, but that sounds like it'll be bad for
 performance?



 Thanks again





 No takers? :(

 *[Danny Nicholas] *

 *In your original scenario you were opening yourself to probable latency
 issues – I would personally pursue something along the line of option B
 where I put the DB data into a temp file and ran a daemon to clear the temp
 files hourly or daily as needed.  If the delivery worked well across most
 LAN’s/WAN’s, some gung-ho developer would have hosed another part of
 Asterisk trying to get that “bell and whistle” into the trunk.*

 Thanks Danny. I'm not so sure, that latency will be that much of an issue
being on the same physical GbE switch as the DB server without any other
traffic on it but sure, I know that a long time ago when I implemented
Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess
we do need to use that AGI AddOn of PUT SOUNDFILE after all.

Would be good if more people can throw a few ideas around to see if there's
a smarter way to do it. Another idea we had was to dumb these files (since
they'll be very small in duration and thus in size) into a directory, run a
web-server and have AGI retrieve them using curl and just use Background
to play it. Thoughts?
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Re: [asterisk-users] Retrieving/Streaming audio/video files from DBusing over AGI

[A E [Gmail]]

On Mon, May 2, 2011 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote:

--

  *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
 *Sent:* Monday, May 02, 2011 1:23 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Retrieving/Streaming audio/video files
 from DBusing over AGI





  Just realised that this can better be described another way:



 What we're essentially trying to do is be able to do any one of these



 a) stream an audio/video file stored in the DB via AGI into the current
 channel so that it plays on the phone



 OR



 b) Do something like what Realtime Voicemail does, where it gets the file
 from the DB, saves as a temp file in the user mailbox directory and then
 plays it to the caller but this needs to happen through AGI, something along
 the lines of readsql (a la func_odbc) inside of AGI



 OR



 c) Anything else that's better than a) and b) above that someone can
 suggest.



 P.S I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which
 seems to be the only solution we can think of right now, other than of
 course having the DB machine exporting the SAN volume as an NFS share for
 the Asterisk server to mount, but that sounds like it'll be bad for
 performance?



 Thanks again





 No takers? :(

 *[Danny Nicholas] *

 *In your original scenario you were opening yourself to probable latency
 issues – I would personally pursue something along the line of option B
 where I put the DB data into a temp file and ran a daemon to clear the temp
 files hourly or daily as needed.  If the delivery worked well across most
 LAN’s/WAN’s, some gung-ho developer would have hosed another part of
 Asterisk trying to get that “bell and whistle” into the trunk.*



 Thanks Danny. I'm not so sure, that latency will be that much of an issue
 being on the same physical GbE switch as the DB server without any other
 traffic on it but sure, I know that a long time ago when I implemented
 Realtime Voicemail, it worked pretty good, so I'll be happy with b). I guess
 we do need to use that AGI AddOn of PUT SOUNDFILE after all.



 Would be good if more people can throw a few ideas around to see if there's
 a smarter way to do it. Another idea we had was to dumb these files (since
 they'll be very small in duration and thus in size) into a directory, run a
 web-server and have AGI retrieve them using curl and just use Background
 to play it. Thoughts?

 *[Danny Nicholas] *

 *IMO, adding curl to the mix is just going to introduce another possible
 point of failure.  If they are that small, why not do a daemonized delivery
 system?*


By daemonized delivery system, I'm assuming you mean have some background
process running to transport these files from the DB to the asterisk server
and play them?

There are two issues with that
a) Sounds like too much I/O esp. with small files getting written and
deleted.

b) What if there are several asterisk servers and the call can come into any
of the servers. Do we invoke the daemon at will, run a SQL query, extract it
from the DB, and transfer it to the asterisk server which initiated the
request and then play it? Sounds like it might add a bit more latency than
streaming it right inside the connection opened by AGI itself, although we
could not store these files in the DB and just have them sit on a dedicated
SAN volume and whenver a request comes in, we send it to the requesting
asterisk server.

That's all of course if I understood you correctly.
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Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

[A E [Gmail]]

On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote:

 Just from my experience with different DBs, stay away from BLOB data
 types as much as possible.

 Hi CF,
any particular reason why? I've had a good experience with it, in fact
that's recommended by DB developers when it's a case of small files. They
say only larger files greater than 500K-1MB should be stored on the
filesystem using filestream or similar etc.

Although at this point, this might be a moot point, as so far no one's been
able to suggest a way to be able to stream the content of the BLOB field to
Asterisk over the AGI connection into the current channel, such that
Asterisk can just play it on the fly. We'll have to just go with getting the
file to the requesting * server and then play it
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[asterisk-users] Asterisk behind D-Link ADSL router with private IP

[gmail]

i have this configuration , An Asterisk server connected to my private LAN 
192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call 
from Internet into Asterisk wireshark show the message destintion port 
unrechable 

i configured sip.conf for nat=yes and qualify=yes and externip=my public 
IP 

did i forget some other ports to forward otherthan 5060?

did i forget any other configurations?

i even tried the virtual server function in my D-Link 2640U ADSL router with 
no hope 


appreciate your help -- 
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[asterisk-users] Asterisk or 3CX?

[gmail]

does 3CX compare to Asterisk in anyhow? it is based on windows and it seems 
that it is more easier to configure than Asterisk , however i think the 
complexity of Asterisk configuration comes with its flexibility , am i right?___
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[asterisk-users] video call doesn work

[gmail]

i am trying to make a video call on asterisk 1.6 , my configuration is an
-  asterisk 1.6 on Centos on virtual machine VmWare
-  Xlite softphone one windows xp (the Host operating system)
-  X-lite client on another windows XP (the Guest operating system )

i put the paramtervideosupport=yes   under the general section  in   
sip.conf
i allowed the video codecs for each client in sip.conf for the clients 3500 and 
3501 

i installed 2 web cams one for each client , and in the X-lite video 
side-window each cam operate well on its corresponding X-lite client in the 
down part, and when i start a call from 3500 to 3501 and the call established 
and i press the send video button  on both clients , but the video stream is 
not sent to any of the 2 clients 
what's wrong?
am i missing something? or does the VmWare enviroment cause the problem and i 
need 2 seperate physical machines 

Gres___
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Re: [asterisk-users] video call doesn work

[gmail]

i already did that
  - Original Message - 
  From: Danny Nicholas 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Wednesday, June 24, 2009 1:08 PM
  Subject: Re: [asterisk-users] video call doesn work


  Make sure the video codecs in the xlite setup are also in sip.conf 
(allow=ulaw,alaw,gsm,h263)

   


--

  From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gmail
  Sent: Thursday, June 25, 2009 12:57 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] video call doesn work

   

  i am trying to make a video call on asterisk 1.6 , my configuration is an

  -  asterisk 1.6 on Centos on virtual machine VmWare

  -  Xlite softphone one windows xp (the Host operating system)

  -  X-lite client on another windows XP (the Guest operating system )

   

  i put the paramtervideosupport=yes   under the general section  in   
sip.conf

  i allowed the video codecs for each client in sip.conf for the clients 3500 
and 3501 

   

  i installed 2 web cams one for each client , and in the X-lite video 
side-window each cam operate well on its corresponding X-lite client in the 
down part, and when i start a call from 3500 to 3501 and the call established 
and i press the send video button  on both clients , but the video stream is 
not sent to any of the 2 clients 

  what's wrong?

  am i missing something? or does the VmWare enviroment cause the problem and i 
need 2 seperate physical machines 

   

  Gres



--


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 http://lists.digium.com/mailman/listinfo/asterisk-users___
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[asterisk-users] cant get a x100p works

[Manolet Gmail]

I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic

i want to configure a x100p card an use it with asterisk, so i download,
compile and install:

asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9

i try almost everything i found on the net but without success:

if i run lspci:
04:06.0 Communication controller: Motorola Wildcard X100P

when i run dahdi_hardware appears this:
pci::04:06.0 wcfxo-   1057:5608 Wildcard X100P

if i run dahdi_cfg -v :
DAHDI Tools Version - 2.1.0.2

DAHDI Version: 2.1.0.4
Echo Canceller(s):
Configuration
==


0 channels to configure.

when i run dahdi_scan:
[1]
active=yes
alarms=UNCONFIGURED
description=DAHDI_DUMMY/1 (source: HRtimer) 1
name=DAHDI_DUMMY/1
manufacturer=
devicetype=DAHDI Dummy Timing
location=
basechan=1
totchans=0
irq=0

if i fo dahdigenconf everythink still same. I also reboot and do modprobe
wcfxo. not success...
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[asterisk-users] problema con una x100p

[Manolet Gmail]

Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic

Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
descague compile e instale lo siguiente:

asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9

Sin embargo no logro configurar la tarjeta con exito, ya probe casi  todo.

Esto aparece si ejecuto lspci:
04:06.0 Communication controller: Motorola Wildcard X100P

dahdi_hardware me muestra:
pci::04:06.0 wcfxo-   1057:5608 Wildcard X100P

dahdi_cfg -v :
DAHDI Tools Version - 2.1.0.2

DAHDI Version: 2.1.0.4
Echo Canceller(s):
Configuration
==


0 channels to configure.

dahdi_scan:
[1]
active=yes
alarms=UNCONFIGURED
description=DAHDI_DUMMY/1 (source: HRtimer) 1
name=DAHDI_DUMMY/1
manufacturer=
devicetype=DAHDI Dummy Timing
location=
basechan=1
totchans=0
irq=0

Nada parece funcionar y realmente no se donde esta el error... alguna idea?

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Re: [asterisk-users] cant get a x100p works

[Manolet Gmail]

On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote:
 I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic

 i want to configure a x100p card an use it with asterisk, so i download,
 compile and install:

 asterisk-1.4.24
 dahdi-linux-2.1.0.4
 dahdi-tools-2.1.0.2
 libpri-1.4.9

 i try almost everything i found on the net but without success:

 if i run lspci:
 04:06.0 Communication controller: Motorola Wildcard X100P

 when i run dahdi_hardware appears this:
 pci::04:06.0     wcfxo-       1057:5608 Wildcard X100P

 What's the output of:

  lsmod | grep ^dahdi


r...@lhserver:~# lsmod | grep ^dahdi
dahdi_dummy11620  0
dahdi_transcode15244  1 wctc4xxp
dahdi 202280  13
dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp

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Re: [asterisk-users] problema con una x100p

[Manolet Gmail]

system.conf:

# Global data

loadzone= us
defaultzone = us

el archivo /etc/asterisk/chan/dahdi.conf no existe, estoy usando
asterisk 1.2 de todas maneras como dije ztcfg -v tampoco muestra la
tarjeta, asi que de ninguna forma la va a ver asterisk, si hago en
asterisk dahdi show channels no aparece nada.

2009/4/2 Brandon B. bran...@brellsystems.com:
 nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y
 /etc/asterisk/chan/dahdi.conf



 2009/4/2 Manolet Gmail mano...@gmail.com

 Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic

 Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
 descague compile e instale lo siguiente:

 asterisk-1.4.24
 dahdi-linux-2.1.0.4
 dahdi-tools-2.1.0.2
 libpri-1.4.9

 Sin embargo no logro configurar la tarjeta con exito, ya probe casi  todo.

 Esto aparece si ejecuto lspci:
 04:06.0 Communication controller: Motorola Wildcard X100P

 dahdi_hardware me muestra:
 pci::04:06.0 wcfxo-   1057:5608 Wildcard X100P

 dahdi_cfg -v :
 DAHDI Tools Version - 2.1.0.2

 DAHDI Version: 2.1.0.4
 Echo Canceller(s):
 Configuration
 ==


 0 channels to configure.

 dahdi_scan:
 [1]
 active=yes
 alarms=UNCONFIGURED
 description=DAHDI_DUMMY/1 (source: HRtimer) 1
 name=DAHDI_DUMMY/1
 manufacturer=
 devicetype=DAHDI Dummy Timing
 location=
 basechan=1
 totchans=0
 irq=0

 Nada parece funcionar y realmente no se donde esta el error... alguna
 idea?

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[asterisk-users] Multiple Repeated tones with TDM02B

[Manolet Gmail]

Hi, I got a card from Digium TDM with 2 FXO modules (red ones). There is a
problem that has me quite upset and is that asterisk always detect tones
repeated two, three or more times.

i mean, if i press 123 on my phone. asterisk detects somethin like:

111223
or 112333
or things like that.

How can I fix it? I tried to change the volume level but this happens even
if is set to 0. Why this happens? there any way to make it work well? I have
the same problem with a x100P
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[asterisk-users] Create virtual extension

[Manolet Gmail]

Have, i want to create a sip extension to a context in my dialplan.
how i can do that?

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[asterisk-users] how to make a ip to ip call

[Manolet Gmail]

Hi, i want to make a direct ip to ip call (without a sip proxy), what
software i can use (windows)? i try with xlite but dont understand how

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[asterisk-users] [OFFTOPIC][SPANISH] Crean do una comunidad de asterisk en español

[Manolet Gmail]

Hola a todos, estoy creando una comunidad de asterisk en español que
se dividira en un blog y un foro, estoy buscando gente que quiera
ayudarme a escribir articulos para el blog, y claro, pueda participar
en el foro.

Si a alguien le interesa saber mas escribanme un mail.

[EMAIL PROTECTED]

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[asterisk-users] features.conf not working

[Manolet Gmail]

Hi, im a new user to asterisk. i have configured one box using asterisknow.

now i want to enable *9 (or some code) to play for example tt-monkeys.

i read a lot in voip-info but cant do it:

i have this on my features.conf:

[applicationmap]
testfeature = *9,callee,Playback,tt-monkeys

extensions.conf:

[globals]
DYNAMIC_FEATURES=testfeature
trunk_1 = Zap/g1
trunk_2 = Zap/g2


what else i have to add in order to make this works? im using 2 xlite,
please help me

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[asterisk-users] connecting 2 FXS together

[Joao Ferreira gmail]

Hello all,

this might be a crazy question

can I connect 2 FXS plugs to the same analog phone ?

my reason: I'm expecting that, with this setup, the phone could operate
transparently through the redundant FXS if the main FXS would fail... of
if asterisk is stopped on one of the servers...

the idea is that the users would not even realize one of the asterisks
is not working and the call was routed by the 'spare' asterisk...


|| FXS; asterisk1
PHONE --|shunt   |
|| FXS; asterisk2 (spare)


has anyone tried this ?

thanks in advance

Joao



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[asterisk-users] using cell phone as an FXO port

[gmail]

Hi all,
I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try 
this and configured it and how to physically connect it to Asterisk server?___
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[asterisk-users] PCI ISDN as a PSTN gateway card

[gmail]

Is there any ISDN PCI cards that can be used with Asterisk as a PSTN gateway 
instead of using Diguim FXO cards?___
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[asterisk-users] Asterisk for larg

[gmail]

Does anyone know how to off-load an Asterisk Box so that to distribute its 
functions like IVR and VoiceMail or its PTSN gateway function into different 
servers? in this case , will the installation of Asterisk on each server 
differe and how these different  servers will interact as a single logical -vs 
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[asterisk-users] Asterisk for larg

[gmail]

Does anybody know how to off-load an Asterisk Box so that to distribute its 
functions like IVR and VoiceMail or its PTSN gateway function into different 
servers? in this case , will the installation of Asterisk on each server 
differe and how these different  servers will interact as a single logical -vs 
physical- server? thx alot ___
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Re: [asterisk-users] chan_ss7 0.10

[Joel @ Gmail]

Hi marek,

gr8. I am working on chan_ss7 now..


Regards,
Joel

- Original Message - 
From: marek cervenka [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, March 03, 2008 3:55 AM
Subject: Re: [asterisk-users] chan_ss7 0.10


 Thanks for the update.
 I have Sangoma A104D and wanted to use ss7 signalling. I came accross
 chan_ss7 but found sifira is not in active development.  But is this
 chan_ss7 stable and can be used in production server implementation.
 We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am

 looking for open source ss7 implementation which is chan_ss7. so need to
 know about stability and recommendation for using on production server.
 
 long term supported solution is libss7 from digium. but this depends on 
 asterisk 1.6 which is not officialy stable
 
 chan_ss7 is now developed by www.dicea.dk.
 http://www.dicea.dk/company/downloads
 it's used on production servers. it is very stable solution
 
 ---
 Marek Cervenka
 ===
 
 
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Re: [asterisk-users] chan_ss7 0.10

[Joel @ Gmail]

Hi marek,

Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross 
chan_ss7 but found sifira is not in active development.  But is this 
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am 
looking for open source ss7 implementation which is chan_ss7. so need to 
know about stability and recommendation for using on production server.


Please provide your recommendation  suggestions.


Regards,
Joel
- Original Message - 
From: marek cervenka [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, November 17, 2007 8:49 PM
Subject: [asterisk-users] chan_ss7 0.10


 hi,

 i made tarball with some ss7 patches from www.voip-info.org and other
 places and put this at http://www.freevoice.cz/chan_ss7-0.10.tgz

 Sifira is not in active development anymore :( (but they made good
 work! thanks)

 from Changelog
 New in version 0.10 (community version)
 - port to asterisk 1.4.14 (http://br.geocities.com/bruno_agostinho/)
 - added E prefix for emergency calls (www.tvtrinec.cz)
 - some stability fixes (www.tvtrinec.cz)
 - sangomazaptel example config
 - RBT (?)
 - autoPC+uptime+watermark+stats (www.ss7.pl)
 - cic block/unblock fix (tomasz.paszkowski at ctinf.pl)
 - local/remote hangup info in NOTICE (cervajs at freevoice.cz)

 please test and report
 thanks

 ---
 Marek Cervenka
 ===


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Re: [asterisk-users] chan_ss7 0.10

[Joel @ Gmail]

Hi marek,

Thanks for the update. 
I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7 
but found sifira is not in active development.  But is this chan_ss7 stable and 
can be used in production server implementation. 
We are going to have 2 to 3 boxes with ss7 signalling using sangoma but I am 
looking for open source ss7 implementation which is chan_ss7. so need to know 
about stability and recommendation for using on production server.


Please provide your recommendation  suggestions.


Regards,
Joel
- Original Message - 
From: marek cervenka [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, November 17, 2007 8:49 PM
Subject: [asterisk-users] chan_ss7 0.10


 hi,
 
 i made tarball with some ss7 patches from www.voip-info.org and other 
 places and put this at http://www.freevoice.cz/chan_ss7-0.10.tgz
 
 Sifira is not in active development anymore :( (but they made good 
 work! thanks)
 
 from Changelog
 New in version 0.10 (community version)
 - port to asterisk 1.4.14 (http://br.geocities.com/bruno_agostinho/)
 - added E prefix for emergency calls (www.tvtrinec.cz)
 - some stability fixes (www.tvtrinec.cz)
 - sangomazaptel example config
 - RBT (?)
 - autoPC+uptime+watermark+stats (www.ss7.pl)
 - cic block/unblock fix (tomasz.paszkowski at ctinf.pl)
 - local/remote hangup info in NOTICE (cervajs at freevoice.cz)
 
 please test and report
 thanks
 
 ---
 Marek Cervenka
 ===
 
 
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Re: [asterisk-users] Call Center SoftPhone with Auto Answer

[Leonardo Kamache (Gmail)]

X-Lite do what you need...




On 8/6/07, Joao Pereira [EMAIL PROTECTED] wrote:
 Hello
 I need a Softphone with auto answer where users can't turn it off.
 Does someone knows a softphone where users can't turn the auto answer off?
 Or is there any way Asterisk could force the clients to answer the phone?

 Thanks
 Regards
 Joao Pereira

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Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql

[Leonardo Kamache (Gmail)]

Do you have MySQL installed in your machine???



On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote:




 No one faced a problem like this !!



  


 From: Khaled Chehab [mailto:[EMAIL PROTECTED]
  Sent: Thursday, June 21, 2007 12:37 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Cc: [EMAIL PROTECTED]
  Subject: asterisk 1.4.1 app_addon_sql_mysql




 I am using centos 4.4 updated using yum



 when I enter asterisk-addons-1.4.1  directory and make menuselect

 *


 Asterisk-addons Module Selection


 *




 Press 'h' for help.



 XXX
 1.  app_addon_sql_mysql

 [*]
 2.  app_saycountpl

 XXX
 3.  cdr_addon_mysql

 [ ]
 4.  chan_ooh323

 [*]
 5.  format_mp3

 XXX
 6.  res_config_mysql



 Cannot install app_addon_sql_mysql ….

 Any dependencies required ?





 Regards








  
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Re: [asterisk-users] Transfer caller direct to voicemail

[Leonardo Kamache (Gmail)]

Hello Drew;

Assuming your extensions is 105 let's see the dialplan:

exten = 105,1,Dial(SIP/105,30,Tt)
exten = 105,n,Hangup

exten = *XXX,1,Answer
exten = *XXX,n,VoiceMail(${EXTEN:[EMAIL PROTECTED])
exten = *XXX,n,Hangup

I think this should work for what you want.


Regards;

Leonardo Kamache
Rio de Janeiro - Brasil



On 6/12/07, Drew Gibson [EMAIL PROTECTED] wrote:

Hi,

Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.

My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well with our dialplan.

According to an article on voip-info.org [EMAIL PROTECTED] appears to
implement this as #*XXX. I assume they are using an application map in
features.conf but I cannot see a way to pass the required extension to
the VoiceMail() application.

Can this be done in features.conf?

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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Re: [asterisk-users] Softphone behind NAT issues

[Leonardo Kamache (Gmail)]

In [general] section:

externip=your_extern_ip_address
localnet=your_local_net/bits   i.e. 192.168.0.0/24

Try this...




On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote:


 We are trying to use a softphone from a location that is behind a firewall.
We are using x-lite as the softphone.

 So far, we've been able to get the phone to register with the asterisk
server, and it can receive voice from the asterisk server (IE, voicemail,
etc).

 However, asterisk can't hear anything from the softphone. We have used 2
different machines to test this on. We are watching the traffic and noticed
that there doesn't appear to be any rtp traffic going back to asterisk (this
is where we think the problem might be). The firewalls on both sides have
ports 5060, 1-2 and 3478 (STUN) open.

 Out conf files are:
 --
 [sip.conf]

 [general]
 context=incoming; Default context for incoming calls
 bindport=5060   ; UDP Port to bind to (SIP standard port is
5060)
 bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
 srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
 allow = all

 [1000]
 nat=yes
 type=friend
 secret=Polycom
 context=internal
 host=dynamic
 canreinvite=no
 [EMAIL PROTECTED]
 callerid=TESTUSER1 1000

 -
 [extensions.conf]
 exten = 1000,1,Macro(stdexten,[EMAIL PROTECTED],SIP/1000)
 

 [rtp.conf]

 [general]
 rtpstart=12000
 rtpend=12005
 dtmftimeout=3000

 What are we missing?

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Re: [asterisk-users] Voip-info.org

[Leonardo Kamache (Gmail)]

Yes from Brazil...




On 6/6/07, Ed Nuñez [EMAIL PROTECTED] wrote:





Is anyone else having trouble going into voip-info.org today?
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Re: [asterisk-users] spa 3102 incoming call

[Leonardo Kamache (Gmail)]

Hi Damiano!

Take a look at this link:

http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159lid=6862769263B11


Best regards;

Leonardo Kamache



On 6/5/07, damiano bertuna [EMAIL PROTECTED] wrote:

Hi to everybody,

I have an spa 3102 where i connected an analog phone (in the fxs port) and
the pstn line (in the fxo port).

This is my problem:

the incoming call doesn't arrive to asterisk.

 In the spa web page i configured this dialplane:

(:[EMAIL PROTECTED]:5060)

where line01 is the context in sip.conf, 192.168.1.220 is the asterisk ip
and 5060 is the asterisk sip port.

[line01]
username = usersipura
fromuser = usersipura
secret = pwdsipura
host = 192.168.1.222
fromdomain = 192.168.1.222
port = 5061
type = friend
dtmfmode = rfc2833
context = call_in
insecure = very


Why?
is the dialplane wrong?

help me, please.

Damiano.

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Re: [asterisk-users] Call Pick Up

[Leonardo Kamache (Gmail)]

Two words for you... parking lot.
Try to transfer your call to extension 700 and see what hapens...




On 4/27/07, Jim Duda [EMAIL PROTECTED] wrote:

I use Asterisk in my house.  Each phone is a different extension.  I
really like the ability to have multiple simultaneous calls in the
house.  However, I do miss being able to be able to pick up a phone in a
different room.  Currently, I have to either transfer the call or
transfer the call to a conference extension to move around the house.

While a connection in progress on one extension, I would like to go to
any other phone, dial some extension number, in order to ether pick up
the call or join in an automatic conference.  In other words, make it
work like the old ma bell phone (when I want it to :-) )

Is this possible in Asterisk?

Thanks,

Jim

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Re: [asterisk-users] Transer calls hitting #

[Leonardo Kamache (Gmail)]

Try to configure your PAP2 DTMF send mode to INFO.




On 4/21/07, Doug Lytle [EMAIL PROTECTED] wrote:

Poul Moller wrote:
 Are there any special ATA audio setting I should apply?


That I don't know, I've never setup an ATA before.

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] Why duoble digits must be so fast to activate features?

[Leonardo Kamache (Gmail)]

Hi Mauro;

Try to add featuredigittimeout = 1500 at features.conf in the [global] section.






On 4/20/07, Gordon Henderson [EMAIL PROTECTED] wrote:

On Fri, 20 Apr 2007, Mauro Zanin wrote:

 Hi everybody,

 I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm
 using Trixbox..).
 I must be as fast as a flash to press *2 and do an attended transfer. If I
 wait only a tenth of a second nothing happens.
 I think it is an issue. I have seen the source code and found nothing bad.
 Is this a known issue?

Change it in features.conf.

Gordon
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Re: [asterisk-users] SIP REGISTRATION TIME OUT

[Manolet Gmail]

Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with the xlite have
DMZ).

But now i close xlite and put the same extension on a grandstream 286
(dont have DMZ). When somebody calls me the caller can hear me. but i
cant hear!

whats the problem? with other providers i can talk using my
grandstream 286 without give it dmz or changing the configuration on
my router.

i hopes somebody can help me!

2007/4/14, dave cantera [EMAIL PROTECTED]:

hello,
I use both * 1.4 and *NOW...   because the *gui is incomplete in *NOW, I
loaded 1.4 over *NOW because the gui regenerates files that, well, don't
seem to work very well.  it seems to me the gui creates the users.conf
file, and then a script creates or uses the users.conf to create the
dialplan...  here is the users.conf file from *NOW...

as you can see, this file does not conform to either sip.conf or
extensions.conf, so that is my reasoning that it is source for some
other generator...
daveC

;!
;! Automatically generated configuration file
;! Filename: users.conf (/etc/asterisk/users.conf)
;! Generator: Manager
;! Creation Date: Sun Jan 21 15:41:42 2007
;!
[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
host = dynamic
localextenlength = 4
;[6000]
;fullname = Joe User
;email = [EMAIL PROTECTED]
;secret = 1234
;zapchan = 1
;hasvoicemail = yes
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international





Nicholas Campion wrote:
 The quick way to check if a user is defined is to go to the asterisk
 console and type sip show users which will list all the defined
 users and passwords.

 You say that it isn't a networking issue, but the fact that you are
 behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100)
 is causing the problem (i think).  All of your packets are reaching
 the server, but when it tries to respond it is sending the packets to
 192.168.0.100 http://192.168.0.100 which is (obviously) not what you
 want to happen.  The solution to this (typically) is to add NAT=yes
 to sip.conf in the general section.

 Give that a try and see what your result is.

 Nick

 On 4/13/07, *Alex Balashov* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:


 On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:

  mmm are you sure that asterisk-gui generate it on the sip.conf file?
  cause i see a new file called users.conf, and i can see the sip
 users
  on it. Anybody uses asterisk now and can check it please??

Hmm.  I use 1.4.x here and installed the stock config file samples
 bundle, and there's no trace of users.conf.

But then again, I have never used any GUI configurator, so I'm
 not in the
 best position to know what sort of structure and metadata it
 generates.

 --
 Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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 Checked by AVG Free Edition.
 Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 
08:34 PM


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Re: [asterisk-users] MeetMe Error

[Manolet Gmail]

I use modprobe ztdummy, next i restart asterisk and now works fine,
modprobe is to load the driver rigth? what i need to do in order to
load automatically, not at the boot time but when asterisk start?

2007/4/19, Ronaldo [EMAIL PROTECTED]:

  Hi,

  Check if your system has the /dev/files needed.
  I think some installation didn't do it automatically.


Manolet Gmail wrote:
 2007/4/18, Ronaldo [EMAIL PROTECTED]:
 Hi Manolet,

 You have to install zaptel in order to make MeetMe application to work.
 MeetMe needs a kind of timer device that is provided by zaptel package.
 Eventhough you don't have a zaptel card you need to install its package.

 Search for MeetMe application in http://www.voip-info.org/ and you will
 find documentation about how to do that.

 Good Luck.

 Ronaldo

 Manolet Gmail wrote:
  Hi! i have an error using the meetme aplication, and just dont work..
  my meetme.conf is:
 
  [rooms]
  conf = 700
 
  i calling from a sip phone, the extension number is 600. there is the
  error:
 
  Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58,
  700|MI) in new stack
  WARNING[20055]: channel.c:3024 ast_request: No channel type registered
  for 'zap'
  WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo
  channel - trying device
  WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo
 device
  SIP/600-09111e58 Playing 'conf-invalid' (language 'es')
  Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on
  'SIP/600-09111e58'
 
  i dont have any zap interface. how to solve this?
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 but i have zaptel 1.4.1 installed... there is any special
 configuration or something?
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Re: [asterisk-users] SIP REGISTRATION TIME OUT

[Manolet Gmail]

hi, to get it work i change under sip.conf

nat: route
Allow RTP reinvite:update

with that i can hear, without dmz... but... why?

2007/4/19, Manolet Gmail [EMAIL PROTECTED]:

Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with the xlite have
DMZ).

But now i close xlite and put the same extension on a grandstream 286
(dont have DMZ). When somebody calls me the caller can hear me. but i
cant hear!

whats the problem? with other providers i can talk using my
grandstream 286 without give it dmz or changing the configuration on
my router.

i hopes somebody can help me!

2007/4/14, dave cantera [EMAIL PROTECTED]:
 hello,
 I use both * 1.4 and *NOW...   because the *gui is incomplete in *NOW, I
 loaded 1.4 over *NOW because the gui regenerates files that, well, don't
 seem to work very well.  it seems to me the gui creates the users.conf
 file, and then a script creates or uses the users.conf to create the
 dialplan...  here is the users.conf file from *NOW...

 as you can see, this file does not conform to either sip.conf or
 extensions.conf, so that is my reasoning that it is source for some
 other generator...
 daveC

 ;!
 ;! Automatically generated configuration file
 ;! Filename: users.conf (/etc/asterisk/users.conf)
 ;! Generator: Manager
 ;! Creation Date: Sun Jan 21 15:41:42 2007
 ;!
 [general]
 ;
 ; Full name of a user
 ;
 fullname = New User
 ;
 ; Starting point of allocation of extensions
 ;
 userbase = 6000
 ;
 ; Create voicemail mailbox and use use macro-stdexten
 ;
 hasvoicemail = yes
 ;
 ; Create SIP Peer
 ;
 hassip = yes
 ;
 ; Create IAX friend
 ;
 hasiax = yes
 ;
 ; Create H.323 friend
 ;
 ;hash323 = yes
 ;
 ; Create manager entry
 ;
 hasmanager = no
 ;
 ; Remaining options are not specific to users.conf entries but are general.
 ;
 callwaiting = yes
 threewaycalling = yes
 callwaitingcallerid = yes
 transfer = yes
 canpark = yes
 cancallforward = yes
 callreturn = yes
 callgroup = 1
 pickupgroup = 1
 host = dynamic
 localextenlength = 4
 ;[6000]
 ;fullname = Joe User
 ;email = [EMAIL PROTECTED]
 ;secret = 1234
 ;zapchan = 1
 ;hasvoicemail = yes
 ;hassip = yes
 ;hasiax = no
 ;hash323 = no
 ;hasmanager = no
 ;callwaiting = no
 ;context = international





 Nicholas Campion wrote:
  The quick way to check if a user is defined is to go to the asterisk
  console and type sip show users which will list all the defined
  users and passwords.
 
  You say that it isn't a networking issue, but the fact that you are
  behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100)
  is causing the problem (i think).  All of your packets are reaching
  the server, but when it tries to respond it is sending the packets to
  192.168.0.100 http://192.168.0.100 which is (obviously) not what you
  want to happen.  The solution to this (typically) is to add NAT=yes
  to sip.conf in the general section.
 
  Give that a try and see what your result is.
 
  Nick
 
  On 4/13/07, *Alex Balashov* [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:
 
 
  On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
 
   mmm are you sure that asterisk-gui generate it on the sip.conf file?
   cause i see a new file called users.conf, and i can see the sip
  users
   on it. Anybody uses asterisk now and can check it please??
 
 Hmm.  I use 1.4.x here and installed the stock config file samples
  bundle, and there's no trace of users.conf.
 
 But then again, I have never used any GUI configurator, so I'm
  not in the
  best position to know what sort of structure and metadata it
  generates.
 
  --
  Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  
 
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  No virus found in this incoming message.
  Checked by AVG Free Edition.
  Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 
08:34 PM
 

 --
 Building Strong Relationships w/ Intelligent Customer Service
 --

 Interlocking Business Solutions, LLC
 856-380-0894 x5000


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