[asterisk-users] MeetMe and dynamic_features

[Leandro Dardini]

Hello,
I am trying to use a dynamic_features during a MeetMe conference without
any luck. The dynamic_features defined macro works great during a normal
call, but is ignored while on a MeetMe conference.

extensions.conf
[macro-RaiseHand]
exten => s,1,DumpChan(1)

features.conf
RaiseHand => #5,peer/caller,Macro(RaiseHand)

extensions.ael
Set(DYNAMIC_FEATURES=RaiseHand);
MeetMe(1234,F);

I have tried with and without the F parameter...

Any suggestion?

Leandro
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[asterisk-users] Spandsp updated

[Leandro Dardini]

I just noticed there is some sort of new spandsp library.
http://www.soft-switch.org/downloads/spandsp/snapshots/

The version reported was still 0.0.6 and there is absolutely no "whats new"
file.

Is there anyone with more details?

Leandro
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[asterisk-users] Pound and hash

[Leandro Dardini]

Hello,
am I wrong or the audio file for vm-rec-name in en_GB package says "pound"
instead of "hash"?

Pound should be for American while British use hash for the # key.

Leandro
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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Leandro Dardini]

Unfortunately the only log messages regarding that channel are the "joined"
and the "left" for both legs.

VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>

2016-09-17 0:39 GMT+02:00 Max Grobecker :

> Hi,
>
> OK, then it looks like the client transferred the call anywhere else.
> Do you see an entry in your log that refers to the bridge ID
> 00bd58c3-3bce-4f1b-9d79-11eb96f37260 ?
> If there was a transfer, the call *may* have been bridged with the
> transfer destination. Also, the destination might be external,
> so you may see a second call starting at the time where the client left
> the bridge.
>
> Max
>
>
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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Leandro Dardini]

No, there is no Music On Hold starting and the bad thing is the call
duration reported by asterisk was just few seconds while the call duration
reported by the provider was few thousand seconds, the max allowed. So they
will be able to terminate the call on the asterisk side and have it run on
the provider side.

Leandro

2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobec...@ml.grobecker.info>:

> Maybe the client just put the call on hold.
> So the call technically has not ended AND the client does not need to send
> or handle any RTP data.
> Is there any mention of "music on hold" for this channel?
>
> Greetings
>  Max
>
>
> - Nachricht von Leandro Dardini <ldard...@gmail.com> -
>  Datum: Thu, 15 Sep 2016 18:06:14 +0200
>Von: Leandro Dardini <ldard...@gmail.com>
> Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>Betreff: [asterisk-users] Tricking asterisk to think the call has
> ended, but it was continuing on the other side
> An: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>
>
> I am banging my head over a simple asterisk trick I was seeing on one
>> asterisk server.
>>
>> An extension dials an international premium number, the called number
>> answers, then the extension hangups, but the call continue to run on the
>> international number side, generating an high profit for the premium
>> number
>> company and a big loss for the asterisk owner.
>>
>> I think some sort of "transfer" takes place, but I can't identify how they
>> do it and most important, how to prevent it.
>>
>
> - Ende der Nachricht von Leandro Dardini <ldard...@gmail.com> -
>
>
>
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[asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Leandro Dardini]

I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.

An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.

Here the relevant logs:

[2016-09-08 21:00:25] VERBOSE[18771][C-066c] pbx.c: Executing
[0021628990XXX@dialoutbound:595] Dial("SIP/201-boxoffice-0f66",
"SIP/0021628990XXX@SBC002_VirginMedia,60,T") in new stack
[2016-09-08 21:00:25] VERBOSE[18771][C-066c] app_dial.c: Called
SIP/0021628990XXX@SBC002_VirginMedia
[2016-09-08 21:00:27] VERBOSE[18771][C-066c] app_dial.c:
SIP/SBC002_VirginMedia-0f67 answered SIP/201-boxoffice-0f66
[2016-09-08 21:00:27] VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:27] VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:28] VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:28] VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>

Any idea?

Leandro
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[asterisk-users] Different cachertclasses setting for different Music on Hold

[Leandro Dardini]

As you know, there is the following settings

[general]
cachertclasses=yes ; use 1 instance of moh class for all users who are
using it,
; decrease consumable cpu cycles and memory
; disabled by default

It allows to use a single instance of MOH for all users. I'd like to have
this setting different for each Music on Hold class.

Is it possible?

Leandro
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Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

[Leandro Dardini]

No. I thank you for all the hard work done and dedication to the project.

Leandro
Il 06/Lug/2016 11:10 PM, "Joshua Colp" <jc...@digium.com> ha scritto:

> Leandro Dardini wrote:
>
>> This is a great news, thank you. I have open the issue,
>> https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
>> relevant files, let me know if you need more info.
>>
>
> And thank you for testing the release candidate so we can ensure the issue
> is fixed before doing the actual release!
>
> --
> 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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Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

[Leandro Dardini]

This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.

Leandro

2016-07-06 21:46 GMT+02:00 Joshua Colp <jc...@digium.com>:

> Leandro Dardini wrote:
>
>> Hello,
>> I'd like to know if anyone of you is finding my same problems using any
>> recent asterisk version, after 13.7 / 13.8  with chan_sip.
>>
>> If I use any recent asterisk version, after just few seconds asterisk
>> completely locks up, stopping processing SIP/UDP packets. Nothing is
>> written in the asterisk log, but if I run "netstat -nap | grep 5060" I
>> see the UDP buffer filled up.
>>
>> If I step back to asterisk 13.2, then all is fine and asterisk is rock
>> solid.
>>
>> I know I should use PJSIP and chan_sip is no more supported, but at this
>> point, if this is the working state of chan_sip, it should be completely
>> removed.
>>
>
> I've responded on the issue but the backtrace you've provided makes it
> appear as though the issue is actually in ODBC, which since chan_sip is
> using it in your deployment it causes it to lock up (why exactly is
> unknown).
>
> Since it's separate you should create a new issue. If you don't want to I
> can do so tomorrow. The complete console output (with debug going to
> console in logger.conf and core set debug 3) as well as the configuration
> would also be useful.
>
> Cheers,
>
>
> --
> 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] Impossible to use any recent asterisk version with chan_sip

[Leandro Dardini]

Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8  with chan_sip.

If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asterisk log, but if I run "netstat -nap | grep 5060" I see
the UDP buffer filled up.

If I step back to asterisk 13.2, then all is fine and asterisk is rock
solid.

I know I should use PJSIP and chan_sip is no more supported, but at this
point, if this is the working state of chan_sip, it should be completely
removed.

Leandro
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[asterisk-users] Registration server with PJSIP

[Leandro Dardini]

Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am
facing is the lack of "sipregs". With chan_sip, when an extension
registers, the server where it has registered to is stored in sipregs.

Is there something similar in pjsip? How can I find on which server the
pjsip extension has registered to?

Leandro
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[asterisk-users] Recording barged calls

[Leandro Dardini]

Hi,
I'd like to record the barged call... but whichever leg of the call I try
to barge, my speaking is never recorded using MixMonitor. Any idea about
the reason?

Leandro
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[asterisk-users] Manager events when ringing multiple extensions at once and pickupExten is used

[Leandro Dardini]

I run in a weird issue with a BLF application I have written... this
application is just receiving events from Asterisk Manager Interface and
blink the lights accordingly. All almost work perfectly, except when a
pickupexen is used when multiple extensions are dialed.

If extension 105 dials extension 100 and 103 together and extension 104
pickupexten 100, then extension 103 will continue to blink as "ringing"
until the call between 105 and 104 is hangup.

Am I missing something or an event is sent at the wrong time?

Here the events:

105 dials both 100 and 103 using somethign like Dial(SIP-100) ...

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/105-DEVEL-0122
[ChannelState] => 4
[ChannelStateDesc] => Ring
[CallerIDNum] => 105
[CallerIDName] => Charles P. Boyd
[ConnectedLineNum] => 
[ConnectedLineName] => 
[Language] => en
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] => 7654
[Priority] => 1
[Uniqueid] => srv01-1458746336.683
[Linkedid] => srv01-1458746336.683

Extension 100 rings

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/100-DEVEL-0123
[ChannelState] => 5
[ChannelStateDesc] => Ringing
[CallerIDNum] => 100
[CallerIDName] => Brent B. Myer
[ConnectedLineNum] => 105
[ConnectedLineName] => Charles P. Boyd
[Language] => en
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] => sw_14771_RINGALL
[Priority] => 1
[Uniqueid] => srv01-1458746336.684
[Linkedid] => srv01-1458746336.683

Extension 103 rings

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/103-DEVEL-0124
[ChannelState] => 5
[ChannelStateDesc] => Ringing
[CallerIDNum] => 103
[CallerIDName] => Erica V. Watson
[ConnectedLineNum] => 105
[ConnectedLineName] => Charles P. Boyd
[Language] => en
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] => sw_14771_RINGALL
[Priority] => 1
[Uniqueid] => srv01-1458746336.685
[Linkedid] => srv01-1458746336.683

Until now, all perfect... now Extension 104 uses pickupexten on 100

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/104-DEVEL-0125
[ChannelState] => 4
[ChannelStateDesc] => Ring
[CallerIDNum] => 104
[CallerIDName] => Christopher C. Andrews
[ConnectedLineNum] => 
[ConnectedLineName] => 
[Language] => en
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] => *56100DEVEL
[Priority] => 1
[Uniqueid] => srv01-1458746341.686
[Linkedid] => srv01-1458746341.686

Channels are up (105 caller and 104 who pickup)

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/104-DEVEL-0125
[ChannelState] => 6
[ChannelStateDesc] => Up
[CallerIDNum] => 104
[CallerIDName] => Christopher C. Andrews
[ConnectedLineNum] => 
[ConnectedLineName] => 
[Language] => en
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] => *56100DEVEL
[Priority] => 4
[Uniqueid] => srv01-1458746341.686
[Linkedid] => srv01-1458746341.686

[Event] => Newstate
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/105-DEVEL-0122
[ChannelState] => 6
[ChannelStateDesc] => Up
[CallerIDNum] => 105
[CallerIDName] => Charles P. Boyd
[ConnectedLineNum] => 104
[ConnectedLineName] => Christopher C. Andrews
[Language] => es
[AccountCode] => DEVEL
[Context] => ExecHuntList
[Exten] => sw_14771_RINGALL
[Priority] => 26
[Uniqueid] => srv01-1458746336.683
[Linkedid] => srv01-1458746336.683

An Hangup is sent for extension 100, the one from which the call was
borrowed

[Event] => Hangup
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/100-DEVEL-0123
[ChannelState] => 5
[ChannelStateDesc] => Ringing
[CallerIDNum] => 100
[CallerIDName] => Brent B. Myer
[ConnectedLineNum] => 105
[ConnectedLineName] => Charles P. Boyd
[Language] => es
[AccountCode] => DEVEL
[Context] => PickupExten
[Exten] => h
[Priority] => 2
[Uniqueid] => srv01-1458746336.684
[Linkedid] => srv01-1458746336.683
[Cause] => 26
[Cause-txt] => Answered elsewhere

But not to extension 103 ... remind both 100 and 103 were ring together.

In the mean time Extension 104 hangup

[Event] => Hangup
[Privilege] => call,all
[SystemName] => srv01
[Channel] => SIP/104-DEVEL-0125
[ChannelState] => 6
[ChannelStateDesc] => Up
[CallerIDNum] => 104
[CallerIDName] => Christopher C. Andrews
[ConnectedLineNum] => 105
[ConnectedLineName] => Charles P. Boyd
[Language] => es
[AccountCode] => DEVEL
[Context] => authenticated
[Exten] =>

Re: [asterisk-users] Crash asterisk res_odbc

[Leandro Dardini]

Which operating system are you using? I have experienced the same problem
on several OS except for CentOS 6. I suppose an ODBC problem on newer OS
version.

Leandro
Il 24/Feb/2016 05:30 PM, "Maxime"  ha scritto:

> Dear list,
>
> i have a issue
>
> Asterisk crash (Module res_odbc exactly) after the same log who is 
> "*ERROR[23805]
> astobj2.c: bad magic number...*"
> you will see on the log :
>
> Today
>
> [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number
> 0x552f302e for 0x7fe3505b3958*
> [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on
> a x86_64 running Linux on 2013-04-09 14:16:57 UTC
> [2016-02-24 16:00:44] NOTICE[31321] loader.c: 2 modules will be loaded.
> [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Connecting asterisk
> [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: res_odbc: Connected to
> asterisk [MySQL-asterisk]
> [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Registered ODBC class
> 'asterisk' dsn->[MySQL-asterisk]
>
> Yesterday :
>
> [2016-02-23 15:59:12] ERROR[19824] *astobj2.c: bad magic number 0x20 for
> 0x27a5558*
> [2016-02-23 15:59:18] Asterisk 11.2-cert1 built by root @ Voice_server on
> a x86_64 running Linux on 2013-04-09 14:16:57 UTC
> [2016-02-23 15:59:18] NOTICE[23791] loader.c: 2 modules will be loaded.
> [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Connecting asterisk
> [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: res_odbc: Connected to
> asterisk [MySQL-asterisk]
> [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Registered ODBC class
> 'asterisk' dsn->[MySQL-asterisk]
>
> Effect : many trunk sip are down during few minutes
> Oddness : same hours
>
> On google i found many times  "memory corruption was the assumption" ...
>
> Have you ever seen this kind of problem ?
>
> thank you in advance
>
> Version : Asterisk 11.2-cert1
> Os : Debian 7-64
>
> --
>
> Maxime
>
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Re: [asterisk-users] Best Asterisk Platform

[Leandro Dardini]

Please chech also MiRTA PBX http://www.mirtapbx.com ... it is a multitenant
realtime multiserver interface.

Leandro
Il 23/Dic/2015 09:06 AM, "er ic"  ha scritto:

> Although, I do like the OS information. I personally am a fan of CentOS.
>
> I realize now that the platform was ambiguous.
>
> On Wed, Dec 23, 2015 at 10:04 AM, er ic  wrote:
>
>> correct, PBX Manager
>>
>> On Wed, Dec 23, 2015 at 9:55 AM, Tech Support 
>> wrote:
>>
>>> I don’t think the original poster was asking about which OS is best. I
>>> think he was asking which PBX manager people are using. Ex, PBX in a Flash,
>>> Elastix, FreePBX, blah, blah, blah.
>>>
>>> Thanks;
>>>
>>> John
>>>
>>>
>>>
>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *John Novack
>>> *Sent:* Wednesday, December 23, 2015 9:24 AM
>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>> *Subject:* Re: [asterisk-users] Best Asterisk Platform
>>>
>>>
>>>
>>> Are you trying to start a religious argument?
>>>
>>> CentOS /RedHat appear to be the most trouble free when compiling from
>>> source
>>> JMO
>>>
>>> John Novack
>>>
>>> er ic wrote:
>>>
>>> What is the best asterisk platform to use? What are you guys using?
>>>
>>> I am looking for something to host either in our data center or at the
>>> customer prem where I have the control over the unit and not through a
>>> contractor.
>>>
>>> I dont mind paying a license fee for a front end interface but still
>>> would rather not have to pay.
>>>
>>> Thanks,
>>>
>>> --Eric
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> Dog is my Co-pilot
>>>
>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
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Re: [asterisk-users] Network range in trunk definition

[Leandro Dardini]

I see, really thank you ... I have just migrated my config. By the way ...
is pjsip realtime supporting realtime registrations?

Leandro

2015-09-08 21:23 GMT+02:00 Joshua Colp <jc...@digium.com>:

> On 15-09-08 04:21 PM, Leandro Dardini wrote:
>
>> I have some problem finding a smart way to add inbound trunks ip
>> authentication. I don't want to set allowguests=yes
>>
>> Some of my providers just list some IP and I add them like:
>>
>> [provider](!)
>> context=fromoutside
>> type=friend
>> insecure=port,invite
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> canreinvite=no
>>
>> [magrathea1](provider)
>> host=87.238.72.129
>> [magrathea2](provider)
>> host=87.238.72.130
>> [magrathea3](provider)
>> host=87.238.72.131
>> [magrathea4](provider)
>> host=87.238.72.132
>>
>> But some providers are not giving single IP, but networks, like
>> 37.157.52.128/25 <http://37.157.52.128/25> and other also /24
>>
>
> chan_sip requires you to use individual entries like this. The res_pjsip
> functionality in 13 and above allows multiple IPs and ranges in a single
> identify section.
>
> --
> 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] Network range in trunk definition

[Leandro Dardini]

I have some problem finding a smart way to add inbound trunks ip
authentication. I don't want to set allowguests=yes

Some of my providers just list some IP and I add them like:

[provider](!)
context=fromoutside
type=friend
insecure=port,invite
disallow=all
allow=g729
allow=ulaw
allow=alaw
canreinvite=no

[magrathea1](provider)
host=87.238.72.129
[magrathea2](provider)
host=87.238.72.130
[magrathea3](provider)
host=87.238.72.131
[magrathea4](provider)
host=87.238.72.132

But some providers are not giving single IP, but networks, like
37.157.52.128/25 and other also /24

How can I deal with them?

Leandro
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[asterisk-users] Escaping parameter for ODBC function

[Leandro Dardini]

Hello,
I just noticed a weird behavior when using ODBC functions. If the content
of any of the paramter has a "=" inside, then the function is not processed
correctly by asterisk.

Let's take for example the following ODBC function in func_odbc.conf

[LOG_SMS]
dsn=asterisk1,asterisk2
synopsis=Log the route of a SMS
writesql=INSERT INTO
sm_smslogs(sm_te_id,sm_date,sm_direction,sm_sourceraw,sm_destraw,sm_from,sm_to,sm_body,sm_fullresult,sm_response)
values
('${ARG1}',NOW(),'${ARG2}','${ARG3}','${ARG4}','${ARG5}','${ARG6}','${SQL_ESC(${ARG7})}','${SQL_ESC(${ARG8})}','${SQL_ESC(${VAL1})}')

When it is called using:

[2015-08-31 16:35:16] VERBOSE[29562][C-0001] pbx.c: Executing
[103@astsms:37] Set("Message/ast_msg_queue", "ODBC_LOG_SMS(1,ONNET,<
sip:102-de...@devel.mirtapbx.com;transport=UDP>,sip:1...@devel.mirtapbx.com;transport=UDP,102,103,Second
test 4,)=SUCCESS") in new stack

Asterisk interprets the first "=" as assignment. In the debug log I found:

Variable: ODBC_LOG_SMS(1,ONNET,,sip:1...@devel.mirtapbx.com;transport=UDP,102,103,Second test
4,)=SUCCESS

And the ODBC function is not executed.

Is there a way, beside using REPLACE, to avoid this problem?

Leandro
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[asterisk-users] Stopping recordings on all legs

[Leandro Dardini]

Hello,I'd like to use a feature code for stopping recordings. Things are
quite easy when the call is received from the outside or just dialed from
inside to outside, but it can go really crazy when there are blind and
attended transfer going on. It ends I don't know on which call leg is the
recording started, so I cannot stop the recording on the right one.
I usually use the following features.conf
#FeatureName =
DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]]
FromOutsideStopMixMonitor =
#0,peer/callee,Macro(pause-recording)FromOutsideStartMixMonitor =
#1,peer/callee,Macro(unpause-recording)
FromInsideStopMixMonitor =
#0,self/caller,Macro(pause-recording)FromInsideStartMixMonitor =
#1,self/caller,Macro(unpause-recording)
So if the call is coming from inside, I use the FromInside, while if the
call is coming from outside, I use the FromOutside in DYNAMIC_FEATURES.
I can use both for the ActivatedBy, but I want also to run the
pause-recording on both channel legs because I do not know on which one the
recording has been started. How can I do?
Here the macros used:
[macro-pause-recording]exten = s,1,NoOp(Stopping Recording -
MIXMONITOR_FILENAME is ${MIXMONITOR_FILENAME})exten = s,n,StopMixMonitor()
[macro-unpause-recording]exten = s,1,NoOp(Resuming Recording -
MIXMONITOR_FILENAME is ${MIXMONITOR_FILENAME})exten =
s,n,MixMonitor(${MIXMONITOR_FILENAME},ab)ldardiniNewsterisk Leandro
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[asterisk-users] Realtime peers and mailbox not existant

[Leandro Dardini]

Some time to time, usually after an asterisk restart or a sip reload, some
realtime sip peers are loaded in memory without their mailbox. I was not
able to replicate the issue on a constant basis, but after adding some
additional logs to asterisk, it seems the add_peer_mailboxes is run
correctly, but then, when the SIP SUBSCRIBE arrives, the mailbox is not
found. If I run a SIP SHOW PEER, the peer is shown without the mailbox.

Have you ever noticed a similar behavior?

Leandro
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[asterisk-users] Realtime peers, mailbox and MWI problem

[Leandro Dardini]

Hello,
I am facing a problem I can't understand. I have several realtime SIP peers
and from time to time, the mailbox field is not loaded in asterisk memory.
The mailbox field is correctly populated in the database, but often, after
an asterisk restart, the mailbox is not associated to the peer (just to
understand, if I run sip show peer 104-TEST, I see the Mailbox empty. If
I run the sip show subscriptiona, I don't see any subscription for the
MWI but only for the BLF.

Is there anyone facing the same problem? How have you solved it?

leandro
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[asterisk-users] Mixing HASH() and LOCAL()

[Leandro Dardini]

The HASH function is really useful when you have to deal with values loaded
using func_odbc, but how do you use with the LOCAL function? Is it possible
to define a HASH as LOCAL?

Leandro
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[asterisk-users] Realtime followme and channel variables

[Leandro Dardini]

Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
under, like instead happen for example in the queue with the
setinterfacevar field.

I just need to pass a variable from the channel placing the call to the
followme to the channel where the extension is dialed by followme. Any idea?

Leandro
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[asterisk-users] Dialing multiple channels with confirm

[Leandro Dardini]

I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.

Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is easy by using the dial
option U(...), but if I dial two extensions at once, when the first
answers, the other stops ringing.

Any idea to make the first continue to ring until the other accept the call?

Leandro
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[asterisk-users] Weird callerid when getting call from Parking lot

[Leandro Dardini]

Hello,
I am experiencing a weird problem on asterisk when I place an outbound
call, park it and then retrieve it. I am using extensions.ael with macro
and switch and I get something as SW_456_... that is autogenerated by
asterisk when compiling the extensions.ael

This doesn't happen when the call comes from outside.

The bad CallerID is displayed only on Cisco 504G phones and it is
transmitted as a Remote-Party-ID

Is there anyone else also getting this bad behavior?

Leandro
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[asterisk-users] Inline transfer

[Leandro Dardini]

Hello,
while most of the physical phones have keys to handle attended and blind
transfer, most soft phones have no support for it. Asterisk offers a
featuremap to assign a key to blindxfer and atxfer and they work fine if
the call is still in the same starting context, but if the call has moved
in another context, then the new call will be started from such context
with unpredictable results.

Do you have any idea to make all transfers to be applied to the context
defined in the sip.conf instead of the context where the call is running in
that moment?

Leandro
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[asterisk-users] Mailbox password change problem on realtime engine

[Leandro Dardini]

Hello,
I am struggling with what seems a common unresolved problem, changing the
password from voicemailman when using a realtime engine (adaptive_odbc in
my case, connected to mysql).

I have seen messages dating back to 2007 with this problem and the last one
was bug 5168, reported as closed, but without explaining the fix

https://issues.asterisk.org/jira/browse/ASTERISK-5168?jql=text%20~%20%22voicemail%20password%22

Just to avoid confusion, I do have a uniqueid column and that is primary
key with auto increment.

I checked the mysql log and no attempt is made to change the password.

Any idea about the source of the problem?

This is my voicemail table:

CREATE TABLE IF NOT EXISTS `voicemail` (
  `uniqueid` int(11) NOT NULL AUTO_INCREMENT,
  `te_id` int(11) NOT NULL,
  `context` char(80) COLLATE utf8_unicode_ci NOT NULL DEFAULT 'default',
  `mailbox` char(80) COLLATE utf8_unicode_ci NOT NULL,
  `password` char(80) COLLATE utf8_unicode_ci NOT NULL,
  `fullname` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `email` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `pager` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `attach` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `attachfmt` char(10) COLLATE utf8_unicode_ci DEFAULT NULL,
  `serveremail` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `language` char(20) COLLATE utf8_unicode_ci DEFAULT NULL,
  `tz` char(30) COLLATE utf8_unicode_ci DEFAULT NULL,
  `tzbytenant` varchar(10) COLLATE utf8_unicode_ci NOT NULL,
  `deletevoicemail` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `saycid` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `sendvoicemail` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `review` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `tempgreetwarn` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `operator` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `envelope` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `sayduration` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `saydurationm` int(3) DEFAULT NULL,
  `forcename` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `forcegreetings` char(3) COLLATE utf8_unicode_ci DEFAULT NULL,
  `callback` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `dialout` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `exitcontext` char(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `maxmsg` int(5) DEFAULT NULL,
  `volgain` decimal(5,2) DEFAULT NULL,
  `imapuser` varchar(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `imappassword` varchar(80) COLLATE utf8_unicode_ci DEFAULT NULL,
  `stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE
CURRENT_TIMESTAMP,
  `welcomeoption` varchar(255) COLLATE utf8_unicode_ci DEFAULT NULL,
  `category` varchar(255) COLLATE utf8_unicode_ci NOT NULL,
  `fromstring` varchar(255) COLLATE utf8_unicode_ci DEFAULT NULL,
  `minsecs` int(11) NOT NULL,
  PRIMARY KEY (`uniqueid`)
) ENGINE=MyISAM  DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci
AUTO_INCREMENT=218 ;
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Re: [asterisk-users] Showing sip subscriptions in Manager

[Leandro Dardini]

The output of the Sip show subscriptions is a formatted text with columns
cut to fit in the page. It can be better than nothing, but I really
dislike to parse it and show incomplete data.

Leandro

2015-01-16 0:03 GMT+01:00 Alex Epshteyn a...@thirdlane.com:

 You can use Command command, and sip show subscriptions as a parameter

 --

 Alex Epshteyn
 email: a...@thirdlane.com
 web: www.thirdlane.com
 phone +1 415.261.6601


 - Original Message -
  From: Leandro Dardini ldard...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
  Sent: Thursday, January 15, 2015 3:00:30 PM
  Subject: [asterisk-users] Showing sip subscriptions in Manager
 
 
 
  Hello,
  almost any useful CLI command has an analogue on Asterisk Manager
  Interface, but I cannot find a way to get the list of subscriptions
  using AMI. Which is the command, if any? The CLI command is sip
  show subscriptions
 
 
  Leandro
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[asterisk-users] Showing sip subscriptions in Manager

[Leandro Dardini]

Hello,
almost any useful CLI command has an analogue on Asterisk Manager
Interface, but I cannot find a way to get the list of subscriptions using
AMI. Which is the command, if any? The CLI command is  sip show
subscriptions

Leandro
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[asterisk-users] Propagating channel driver flag

[Leandro Dardini]

Starting with asterisk 1.8, when you dial multiple channels at once and one
of them is answered, all other channels were canceled with the cause 200 -
Call completed elsewhere, so modern phones don't display the call as
missed.

Do you know a way to transmit this cause over multiple channels? Let me
make an example:

Extension 103 dials an hunt group dialing the extensions 104 and 105, so in
the code I have something:

Dial(SIP/104SIP/105);

When a call is received for 104, a new dial is made for extension 106. If
the call is pickup by extension 105, the cause 200 - Call completed
elsewhere is sent over the channel for 104, but that is not transmitted to
106.

Is it a way to make it happen?

Leandro
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[asterisk-users] SLA (Shared Line Appearance) and realtime

[Leandro Dardini]

Hello,
do you know if it is possible to define the SLA configuration in the
database for realtime usage with asterisk?

Leandro
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[asterisk-users] SPA504G auto answer

[Leandro Dardini]

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging). I
have tried the following SIP headers (not all together), but without luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);

Any other ideas?

Leandro

PS
I have set the Auto Answer Page to yes
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[asterisk-users] Asterisk 12.6 and MWI, no more working

[Leandro Dardini]

Hello,
while moving from asterisk 12.3 to asterisk 12.6, I see the MWI support for
voicemail has stopped working. If I check sip show peer 104-DEVEL on
asterisk 12.3, I can clearly see the Mailbox option set, while on
asterisk 12.6 it appears empty.

Is there anything to do more for having MWI to work on asterisk 12.6? I
just moved the configuration used for asterisk 12.3 to the one running
asterisk 12.6

Leandro
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[asterisk-users] Voicemail message number off by one when using ODBC storage

[Leandro Dardini]

Hello,
have you noticed the message num (VM_MSGNUM) is off by one?

For example, I receive the following message:

Just wanted to let you know you were just left a 0:03 long message (number
7)

but in attach there is the msg0006.wav

Leandro
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Re: [asterisk-users] features.conf and mixmonitor stop and start

[Leandro Dardini]

Can you post an example?

Leandro


2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:

 Do the pause/unpause in a Macro or Gosub and reference that from the
 features.conf

 Also, make sure you put the filename into a variable and give it full
 inheritance so you can resume recording to the same file (using the a
 option)


 On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 I have a recording started in the dialplan with the MixMonitor
 application. I want to be able to stop it during a call and maybe restart
 it.

 I tried using the value defined in [featuremap] but it starts another
 MixMonitor application even if there already one instead of stopping it.

 Any idea on how I can stop the MixMonitor application while it is running?

 [featuremap]
 automixmon = *1

 I tried also to use the [applicationmap]] but it doesn't seem to work.
 Pressing #1 do nothing. Here my dialplan:

 = {
 Set(__DYNAMIC_FEATURE=pauseMonitor);
 MixMonitor(test);
 Dial(SIP/1000@srv01,30,TtX);
}


 [applicationmap]
 pauseMonitor   = #1,self/both,stopMixMonitor

 Any advice?





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 --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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[asterisk-users] features.conf and mixmonitor stop and start

[Leandro Dardini]

Hello,
I have a recording started in the dialplan with the MixMonitor application.
I want to be able to stop it during a call and maybe restart it.

I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.

Any idea on how I can stop the MixMonitor application while it is running?

[featuremap]
automixmon = *1

I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:

    = {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
   }


[applicationmap]
pauseMonitor   = #1,self/both,stopMixMonitor

Any advice?
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[asterisk-users] Calls to voicemail drops after 41 seconds due to no rtp packets

[Leandro Dardini]

Hello,
I have my provider dropping the calls after 41 seconds of not receiving any
RTP from my asterisk. Obviously there is no RTP back when the caller is
leaving a message in the voicemail. Is it possible to have asterisk
generate some RTP packet back?

Leandro
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Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Leandro Dardini]

It is the way it works. First the phone sends a REGISTER without any
password. Asterisk answers with a Unauthorized and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.

Leandro


2014-05-14 13:12 GMT+02:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:


 Hello,

 After a small break from working on this, I got the idea of tcpdumping the
 correct ports. What I see is REGISTER messages from Kamailio port to
 Asterisk, which are replied with 401 Unauthorized. Why is this happening?
 In my sippeers table the secret field has no value (tried both NULL and
 empty string) and the added field sippasswd has the correct password for
 the user.

 The above might be the cause of my problem, would anyone be able to advice
 me to get to correct behaviour? Now Kamailio sees the clients as
 registered, which would be wrong if Asterisk doesn't.

 cheers,
 Olli



 2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
 :


 Hello all,

 I've been testing a Kamailio Asterisk Realtime integration, and found a
 strange situation.

 My problem is that when using the integration, everything seems ok but
 Asterisk does not see the clients as registered. Kamailio and the clients
 report registered clients. Also calls fail.

 In Asterisk cli sip show peers shows nothing but for example realtime
 load sipusers name 660 shows the user data. Field regseconds has a value
 and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as
 they are on the same machine).

 I have a very simple dialplan:

 [general]

 [default]
 exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
  same = n,Dial(SIP/${EXTEN},3600,rt)
  same = n,Hangup


 Here's more on my problem and background to it, guys on the Kamailio list
 helped out but looks like I need to check my Asterisk configuration.
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html

 My goal is to have all clients in the asterisk database, asterisk (one at
 this point, several later) handling the calls and Kamailio as proxy. In
 Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
 domain 'testers.com'.

 I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
 the same rental virtual server. Clients are in my home network behind nat.
 In MySQL I have database asterisk with table sippeers, where I have
 clients added like this:
 INSERT INTO sippeers
 (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
 VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
 ','660','friend');

 In this message there are some outputs and a sip trace of a register:
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html

 What I don't know is how to configure sip.conf, so far I've just been
 making guesses based on online examples and documentation.
 My current sip.conf looks like this:

 [general]
 bindport = 5070
 bindaddr = 127.0.0.1
 tcpbindaddr = 127.0.0.1:5070
 tcpenable = no
 limitonpeers = yes
 ;rtcachefriends = yes
 tos_sip=cs3
 tos_audio=ef
 realm = testers.com

 I've tried defining realm and domain values, but I lack proper
 understanding of those. Can you guys help me out? Are there any other
 configurations I need to check?

 Respectfully,
 Olli




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[asterisk-users] 302 Moved Temporarily and channel variable

[Leandro Dardini]

When a call is transferred to another extension using a blind transfer,
asterisk keeps traces of who is transferring in the BLINDTRANSFER variable.
If instead the call is forwarded using most phone call forward feature, a
302 Moved Temporarily is sent back to asterisk

-- Called SIP/104-DEVEL
-- Got SIP response 302 Moved Temporarily back from
83.211.***.***:5063
-- Now forwarding SIP/103-DEVEL- to
'Local/0039*@authenticated' (thanks to SIP/104-DEVEL-0001)

Unfortunately I don't find in any of the variables available in the Local
channel used by asterisk to place the new call, the originating extension.

In the logs asterisk says Thanks to SIP/104-DEVEL... but in which
variable can I find this value?

Leandro
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Re: [asterisk-users] Strange incoming call issue.

[Leandro Dardini]

About a call not being hang up for asterisk while the client hang up,
please remember SIP is based on UDP and UDP packets get easily lost... they
are retransmitted but sometime they are lost as the previous...

For the ghost calls, are the SIP port of the phones reachable from the
Internet... maybe it is just someone trying to place some free calls

Leandro


2014-02-12 19:05 GMT+01:00 Mike Diehl mdiehlena...@gmail.com:

 Hi all,

 I've got a customer who's reporting ghost calls. Essentially, the phone
 rings, they pick up, and there's no body there.

 It is NOT one-way audio, and it doesn't happen all the time.

 We use voipmonitor to watch calls, and this is what we saw for the call in
 question:

 | calldate| caller | called | duration | whohanged
 |

 +-++++-+
 | 2014-02-12 09:28:06 | 575xxx | CCD539F38...-1 |   60 | NULL
 |
 | 2014-02-12 09:29:06 | 575xxx | CCD539F38...-2 |1 | NULL
 |

 So, it looks like my customer received a call, which lasted a minute, and
 then they  hung up.  Then their phone rang again, but there was no one
 there.
 Based on what I'm seeing in my log, the first call was never hung up, even
 though both parties claim to have hung up the call.  My logs only indicate
 that the 'h' extension was called once, at 9:29:07

 My question is, how can a call not get hung up when both parties hang up
 the call?  I know that sounds odd, but that's what I'm seeing.

 Any ideas?

 Mike.


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Re: [asterisk-users] SPA112 Won't stay up

[Leandro Dardini]

How long is the registration timeout? If the device is behind a
router/firewall, then you need to set a registration timeout lower than the
state table life in the router/firewall. I usually set my devices to just
2 minutes and it works almost all the time. Most Cisco devices have a very
long timeout of 3600 seconds.

Leandro


2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com:

 Hi all,

 I have an SPA112 that in sitting behind a Ubee cable modem.  The internet
 link is solid, but the device becomes unreachable within a day or so of
 being rebooted.  Then the customer goes to reboot the device, they report
 that all 4 lights are lit.  The ISP reports that the device does respond to
 ping, so it's not completely dead.  I've had the same symptoms with
 SPA303's sitting behind Ubee modems.

 So, is there some configuration setting on the SPA that I can set to make
 this device more stable?

 Mike.

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[asterisk-users] CDR(start) returns nothing in Asterisk 12

[Leandro Dardini]

Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the
${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
correct values. Where is my mistake? Has this function being renamed?

Leandro
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Re: [asterisk-users] CDR(start) returns nothing in Asterisk 12

[Leandro Dardini]

I love you all

:-)

Leandro


2014-02-05 Richard Mudgett rmudg...@digium.com:




 On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.comwrote:

 Hello,
 I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
 the ${CDR(start)} is not returning any data. Other functions, like
 ${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
 correct values. Where is my mistake? Has this function being renamed?


 This was just fixed yesterday.  See
 https://issues.asterisk.org/jira/browse/ASTERISK-23250

 Richard


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Re: [asterisk-users] Parking in Asterisk 12.0.0

[Leandro Dardini]

I have converted the normal Park application and I can only alert you about
the syntax change. I suspect also in the ParkAndAnnounce command, the
parameters are ordered completely different.

Leandro


2014-01-30 Anders Larsson aster...@adev.se:

  Hi

 I'm trying to get the rebuilt parking functionality to work in Asterisk
 12.0.0.

 In Asterisk 11.6.0 I managed to get a call to get parked by adding a
 dynamic feature in features.conf for the DMTF sequence *# which called a
 macro in extensions.conf, which then runned the ParkAndAnnounce
 application, and the call got parked.

 The syntax for ParkAndAnnounce I used was this (I don't want any
 announcement to be played):

 exten = s,n,ParkAndAnnounce(,3600,SIP/100)


 In the new Asterisk-version, the ParkAndAnnounce application gets called,
 but the call isn't parked.

 The only error I can see in the messages file is a DEBUG entry saying that
 the channel failed to join Bridge, like this:

 [Jan 30 21:00:01] DEBUG[7118][C-]: bridge_channel.c:1994
 bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12:
 0x16e3768(SIP/vpn-sbc-0001) failed to join Bridge


 Anyone else that has tried to convert old parking functionality into
 Asterisk 12.0.0 ?



 features.conf:

 parkswitch = *#,callee/caller,Macro(parkswitch)


 extensions.conf:

 [default]
 

 include = parkedcalls

 [macro-parkswitch]
 exten = s,1,ParkAndAnnounce(,,PARKED,SIP/100)


 messages:

 [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847
 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF
 begin '*' received on SIP/at-tcty-ssw-
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4061 __ast_read: DTMF
 begin passthrough '*' on SIP/at-tcty-ssw-
 [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2165
 ast_rtp_update_source: Setting the marker bit due to a source update
 [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847
 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at x.x.x.x:9530
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF
 end '*' received on SIP/at-tcty-ssw-, duration 240 ms
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4005 __ast_read: DTMF
 end accepted with begin '*' on SIP/at-tcty-ssw-
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF
 end passthrough '*' on SIP/at-tcty-ssw-
 [Jan 30 21:00:00] DEBUG[7114][C-]: bridge_channel.c:1174
 bridge_channel_feature: DTMF feature string on
 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*'
 [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847
 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at x.x.x.x:9530
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF
 begin '#' received on SIP/at-tcty-ssw-
 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4054 __ast_read: DTMF
 begin ignored '#' on SIP/at-tcty-ssw-
 [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2847
 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at x.x.x.x:9530
 [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF
 end '#' received on SIP/at-tcty-ssw-, duration 230 ms
 [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF
 end passthrough '#' on SIP/at-tcty-ssw-
 [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1174
 bridge_channel_feature: DTMF feature string on
 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*#'
 [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1185
 bridge_channel_feature: DTMF feature hook 0x7f6b8c1d9480 matched DTMF
 string '*#' on 0x7f6b8c10f998(SIP/ssw-)
 [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2165
 ast_rtp_update_source: Setting the marker bit due to a source update
 [Jan 30 21:00:01] DEBUG[7118][C-]: res_rtp_asterisk.c:2165
 ast_rtp_update_source: Setting the marker bit due to a source update
 [Jan 30 21:00:01] DEBUG[7118][C-]: app.c:305 ast_app_exec_macro:
 SIP/vpn-sbc-0001 Original location: default,,1
 [Jan 30 21:00:01] DEBUG[7118][C-]: pbx.c:4875
 pbx_extension_helper: Launching 'ParkAndAnnounce'
 -- Executing [s@macro-parkswitch:1]
 ParkAndAnnounce(SIP/vpn-sbc-0001, ,,PARKED,SIP/100) in new stack
 [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486
 find_best_technology: Bridge technology softmix does not have any
 capabilities we want.
 [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486
 find_best_technology: Bridge technology simple_bridge does not have any
 capabilities we want.
 [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486
 find_best_technology: Bridge technology native_rtp does not have any
 capabilities we want.
 [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:505
 find_best_technology: Chose bridge technology 

[asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

[Leandro Dardini]

When you use a product which version number is 11 or even 12, you might go
with the assumption all big bugs are fixed and then you find there is a
huge, important, expensive bug still running in the code we are relaying
upon...

The problem is simple. If you transfer a call, that dialing will be not
reported in the CDR, so no billing will happen. This is a simple example:

Extension 100 calls extension 101
After 10 seconds, extension 100 transfer the call to
00VERYEXPENSIVEDESTINATION
After 100 seconds, extension 101 hangup the call

What do you find in the CDR? Just one record for a call from extension 100
to extension 101 lasting 10 seconds. What about the 100 seconds call from
100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.

How do you manage these cases?

Leandro
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Re: [asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

[Leandro Dardini]

2014/1/23 Matthew Jordan mjor...@digium.com

 On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini ldard...@gmail.com
 wrote:
  When you use a product which version number is 11 or even 12, you might
 go
  with the assumption all big bugs are fixed and then you find there is a
  huge, important, expensive bug still running in the code we are relaying
  upon...

 First, not all versions in 11 are the same. Bugs do get fixed. What
 version of Asterisk 11 are you using?


I am using asterisk 11.6 and searching for CDR transfer in the issue
tracker return unfixed bugs

https://issues.asterisk.org/jira/browse/ASTERISK-11309
https://issues.asterisk.org/jira/browse/ASTERISK-21822




 Second, CDRs are not the same in Asterisk 12. Due to extensive changes
 in the bridging core, CDRs were re-worked heavily. You may want to
 take a look at the notes on the Asterisk wiki [1] for Asterisk 12, as
 well as the CDR specification for Asterisk 12 [2].


That seems great! Asterisk 12 really solved the CDR problem when
transferring!



  The problem is simple. If you transfer a call, that dialing will be not
  reported in the CDR, so no billing will happen. This is a simple example:

 And how did you do the transfer? Via DTMF features? Via a particular
 channel driver technology? If so, which channel drivers were involved?


Transfer was made using the transfer button of the phone and the result
was the same with blind or attended transfer



 What kind of transfer was it? Blind? Attended? Failed attended (the
 notorious blonde transfer)?

 
  Extension 100 calls extension 101
  After 10 seconds, extension 100 transfer the call to
  00VERYEXPENSIVEDESTINATION
  After 100 seconds, extension 101 hangup the call
 
  What do you find in the CDR? Just one record for a call from extension
 100
  to extension 101 lasting 10 seconds. What about the 100 seconds call from
  100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.
 
  How do you manage these cases?
 

 I'm not sure if there is a bug report filed against CDRs for the
 currently maintained branches for lost records during a blind or
 attended transfer that matches your issue. There is ASTERISK-17826,
 which may or may not be your issue: the noted lack of information
 makes it a bit hard to tell. The last issue that I'm aware of that we
 fixed regarding lost CDRs during a transfer was ASTERISK-21394, which
 was fixed in 11.4.0.

 So, if you're using a version prior to 11.4.0, you may want to
 consider upgrading. Again, due to the lack of information, it's hard
 to tell whether or not that would help you.

 Finally, CDRs in versions of Asterisk prior to 12 are subject to the
 whims of channel masquerades. This has historically made it difficult,
 if not impossible, to guarantee correctness during all transfer
 operations. Additionally, even if we could guarantee a particular set
 of behaviour in all circumstances, the lack of any clear agreement as
 to what a CDR should look like after an attended transfer (or in any
 situation that involved multiple parties) made the problem impossible
 to solve to the satisfaction of everyone. This particular reason is
 why CEL was created. If you continue to have problems with the billing
 records, you may want to consider moving your billing logic to CEL.

 Note that since (a) Asterisk 12 re-architected using a consistent
 bridging framework, which killed visible channel masquerades; and (b)
 we decided to not try and please everyone and just defined CDRs for
 how we thought they should work; the behaviour of CDRs in Asterisk 12
 and in future versions should be substantially more predictable.

 Matt


Thank you a lot! I am going to move ahead with asterisk 12!



 [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12
 [2]
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification

 --
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 Digium, Inc. | Engineering Manager
 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 Fax detection *11.7

[Leandro Dardini]

It is really more interesting the receiving part. Can you paste here?

Leandro


2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

 Hello everybody

 I'm trying to enable the Digium res_fax app at my *11.7 Server.

 a fax show stats comes up with
 FAX Statistics:
 ---

 Current Sessions : 0
 Reserved Sessions: 0
 Transmit Attempts: 0
 Receive Attempts : 1
 Completed FAXes  : 1
 Failed FAXes : 1

 Digium G.711
 Licensed Channels: 1
 Max Concurrent   : 0
 Success  : 0
 Switched to T.38 : 0
 Canceled : 0
 No FAX   : 0
 Partial  : 0
 Negotiation Failed   : 0
 Train Failure: 0
 Protocol Error   : 0
 IO Partial   : 0
 IO Fail  : 0

 Digium T.38
 Licensed Channels: 1
 Max Concurrent   : 1
 Success  : 0
 Canceled : 0
 No FAX   : 0
 Partial  : 0
 Negotiation Failed   : 0
 Train Failure: 1
 Protocol Error   : 0
 IO Partial   : 0
 IO Fail  : 0

 so that should be ok.

 The corresponding dialplan section starts with


 [from-sip]
 include = inbound

 [inbound]
 exten = _X.,1,Answer()
 exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
 exten = _X.,n,Ringing
 exten = _X.,n,Progress()
 exten = _X.,n,Wait(5)
 exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
 ...
 exten = fax,1,NoOp( FAX DETECTED )
 exten = fax,n,Goto(fax-rx,receive,1)

 in the sip.conf i specified

 [general]
 sendrpid=rpid
 trustrpid=yes
 language=de
 videosupport=yes
 callevents=yes
 caninvite=yes
 qualify=yes
 nat=force_rport,comedia
 faxdetect=yes
 t38pt_udptl=yes

 ...

 [abcde]
 type=peer
 insecure=invite
 defaultuser=12345678912
 fromuser=12345678912
 fromdomain=abcde.ab
 secret=guess-what
 host=abcde.ab
 qualify=yes
 context=from-sip
 dtmfmode=rfc2833
 callbackextension=12345678912


 but all i can see if i try to send a testfax is

 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5)
 in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/456,30,oxX) in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP RTP CoS mark 5
 -- Called SIP/200
 -- Called SIP/201
 -- SIP/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing


 Any hints why thats not working?

 Best Regards Jakob


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Re: [asterisk-users] Dialing a SIP URI with an ;ext= parameter

[Leandro Dardini]

I am going to try a Lync server/asterisk integration, so I really
appreciate!

Leandro


2014/1/21 Lincoln King-Cliby linc...@controlworks.com

 Ok, so now I just feel kind of stupid. After I got home I decided to play
 with this a little more.



 After far too long I realized that part of the issue was Asterisk parsing
 the ; as a beginning of a comment (hindsight=duh).

 A little bit more experimenting and (though I could swear I tried this
 before) replacing the ; with \; works.



 That is, to dial a E.164 normalized number with an extension configured as
 tel:+14404491100;ext=1407 +14404491100;ext=1407 with the SIP Peer for
 the Lync mediation server named “lync” the working dial() is



 Dial(SIP/lync/+14404491100\;ext=1407)



 Hope this may save someone else time down the road.



 --

 Lincoln King-Cliby, CTS, DMC-D

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lincoln King-Cliby
 *Sent:* Monday, January 20, 2014 5:04 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Dialing a SIP URI with an ;ext= parameter



 Hi All,



 In the midst of trying to pilot a deployment of Microsoft Lync (mainly for
 non-voice collaboration, specifically IM) and integrate it with our
 Asterisk (11.6.0 if it matters) deployment and a “everything in one place”
 tool when people are out of the office.



 I have everything on the voice side playing  nice from the Lync side
 (Lync-Lync, Lync-Asterisk, Lync-Asterisk-PSTN)  but I can’t get calls
 from Asterisk-Lync passing.



 I think the root issue is Lync demands that the “line URI” be entered in a
 E.164 normalized format, and further specifies that if an extension is
 specified it should be entered as ;ext=. So, e.g. when I have myself set up
 in LYNC my Line URI is entered as 
 “tel:+144044911100;ext=1407+144044911100;ext=1407”.




 If I try feeding that into an Asterisk DIAL() using any format I can think
 of (specific examples below) the call fails and the following is logged to
 console; it looks like Asterisk is dropping the “;ext=”…

   == Using SIP RTP CoS mark 5

 -- Executing [1407@yyy:1] Dial(xx, SIP/lync/
 +14404491100) in new stack

   == Using SIP RTP CoS mark 5

 -- Called SIP/lync/+14404491100

 -- Got SIP response 485 Ambiguous back from IP address and port of
 Lync mediation server

   == Everyone is busy/congested at this time (1:0/0/1)

 -- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL'



 On the other hand, if I change my line URI to a “random” and unused in
 Lync E.164 number without an extension and change the DIAL() to reflect
 that number… the call succeeds, so it seems like I’ve narrowed it down to
 just needing to figure out how to properly pass the extension to Lync.



 The Googling I turned up didn’t seem too positive (and suggested using an
 Exchange Unified Messaging auto attendant and forcing the user to redial
 the extension once connected to the AA was the only alternative for non-DID
 users) but it seems like it should be relatively simple to bridge (what
 seems like a very small) gap.



 Here are the least embarrassing variations on Dial I’ve tried



 Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(“SIP/lync/+14404491100;ext=1407) -- 485 Ambiguous response as above

 Dial(SIP/lync/+14404491100/1407) -- call ‘sits there’ and multiple
 “sip_xmit of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1:
 Invalid argument” logged to console





 Any assistance, is as always very appreciated.



 Thanks!



 Lincoln







 --

 Lincoln King-Cliby, CTS, DMC-D

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



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Re: [asterisk-users] Asterisk Fax detection *11.7

[Leandro Dardini]

I am not sure, but try to add a wait(2) as first command. When I want fax
detection, I insert always a small delay for letting the fax detection
routine to detect it.

Leandro


2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

  Hi

 The log i've posted


 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5)
 in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/456,30,oxX) in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP RTP CoS mark 5
 -- Called SIP/200
 -- Called SIP/201
 -- SIP/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing

 is that what asterisk is showing during an incoming fax call. It looks
 like the faxdetection is not working but why?

 Regards Jakob

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Re: [asterisk-users] Asterisk Fax detection *11.7

[Leandro Dardini]

Please paste the actual code. First has to be the Wait and then any other
thing.

Leandro


2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

  i already added a Progess() and Wait(5) and it still does not detect
 faxes.


 Am 21.01.2014 16:53, schrieb Leandro Dardini:

 I am not sure, but try to add a wait(2) as first command. When I want fax
 detection, I insert always a small delay for letting the fax detection
 routine to detect it.

  Leandro


 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

  Hi

 The log i've posted


 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016,
 5) in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/456,30,oxX) in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP RTP CoS mark 5
 -- Called SIP/200
 -- Called SIP/201
 -- SIP/123-0018 connected line has changed. Saving it until
 answer for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until
 answer for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing

  is that what asterisk is showing during an incoming fax call. It looks
 like the faxdetection is not working but why?

 Regards Jakob

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Re: [asterisk-users] Asterisk ignoring nat settings

[Leandro Dardini]

Yes, thank you. Maybe I have found the problem. The asterisk server is
behind a nat and the RTP port range was not redirected to the asterisk box,
so the Symmetric RTP cannot work because the asterisk is not receiving any
RTP packet from the remote phone.

Leandro


2014/1/16 Ishfaq Malik i...@pack-net.co.uk

 Is directmedia set to no?


 On 15 January 2014 23:11, Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 I have an asterisk box with a peer configured with
 nat=force_rport,comedia, but asterisk keeps sending the audio to the
 private IP address and ignoring the client peer nat settings.

 If I check the sip show peer extension, I see both symmetric RTP and
 Force Rport are set to yes, but asterisk seems ignoring them.

   Force rport  : Yes
   Symmetric RTP: Yes

 Asterisk is behind a nat the the externip and localnet has been
 configured. The local net on the asterisk network is different from the
 local net on phone.

 What else could I check?

 Leandro


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 --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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[asterisk-users] Asterisk ignoring nat settings

[Leandro Dardini]

Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.

If I check the sip show peer extension, I see both symmetric RTP and
Force Rport are set to yes, but asterisk seems ignoring them.

  Force rport  : Yes
  Symmetric RTP: Yes

Asterisk is behind a nat the the externip and localnet has been configured.
The local net on the asterisk network is different from the local net on
phone.

What else could I check?

Leandro
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Re: [asterisk-users] screen capture for asterisk call center solution

[Leandro Dardini]

Just use VNC...


2013/12/20 Goke M Aruna gok...@gmail.com

 Thanks AJ,
 The capturing of agent activities on their desktop by the supervisor.
 Regards
 On 20 Dec 2013 12:18, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Friday 20 December 2013, Goke M Aruna wrote:
  Thank you AJ,
  Just want to know from people who uses asterisk as call center solution,
  how and what screen capture solution / applications are in use.

 What do you mean by screen capture ?

 --
 AJS

 Answers come *after* questions.

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Re: [asterisk-users] [NEWBIE] Right dect to buy to use with asterisk

[Leandro Dardini]

Hello Mario,
nice to meet you on this mailing list!
Gigaset phones are a very high quality/price ratio, so I'll suggest you to
go with the dect ip models. Then you'll need to configure asterisk to act
as IVR, configure a queue and a failover to ring all hunt list.

Drop me a phone call and I'll be happy to help you

Leandro


2013/12/11 Mario Giammarco mgiamma...@gmail.com

 Hello,
 I need to setup this configuration:

 - asterisk as IVR;
 - dect phones.

 So basically I need a standard set of features:

 - each dect phone has its extension so I can call it directly;
 - handover of a call with R key;
 - if a call is not replied by someone ring all phones.

 I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip
 base station.

 Which one should I buy?

 Thanks,
 Mario


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[asterisk-users] Answering agent

[Leandro Dardini]

Hello friends,
when a call arrives in the queue, a CDR record is created, but there is no
info about which agent has picked up the call. I can find that info only in
queue_log.

Is there a way to have that info in the CDR or maybe in a variable in the
h context, when the call is ended?

Leandro
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Re: [asterisk-users] Asterisk 11.6.0 not starting up

[Leandro Dardini]

On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?

Leandro


2013/11/25 Daniel - Asterisk earohua...@gmail.com

 Hello Friends:

 I've just installed Asterisk 11 on my Linux (debian) server but it is not
 starting up when trying with asterisk -vvc and service asterisk
 start. Starting process just stop and shows: Illegal instruction as
 final output.

 Looking at logs I fouind at /var/log/asterisk/messages :

 [Nov 25 11:09:26] Asterisk 11.6.0 built by root @ (my-pbx-server) on a
 i686 running Linux on 2013-11-25 15:10:00 UTC
 [Nov 25 11:09:26] NOTICE[24118] cdr.c: CDR simple logging enabled.
 [Nov 25 11:09:26] NOTICE[24118] loader.c: 205 modules will be loaded.
 [Nov 25 11:09:26] NOTICE[24118] res_odbc.c: res_odbc loaded.
 [Nov 25 11:09:26] NOTICE[24118] res_smdi.c: No SMDI interfaces are
 available to listen on, not starting SMDI listener.
 [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine curl
 [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine odbc
 [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine sqlite3

 Any help would be welcome. My Linux distro is: Linux (my-ip-address)
 3.11.6-x86-linode54 #1 SMP Wed Oct 23 15:22:49 EDT 2013 i686 GNU/Linux

 Elder D. Arohuanca
 Lima - Peru

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[asterisk-users] Dialing directly with username and password

[Leandro Dardini]

It seems I am not finding the right syntax to dial directly using an
username/password. If I insert in my dialplan something like:

12345 = {
  Dial(SIP/823*:5***@78.11.22.33/01342244560);
  hangup();
}

Then I get:

[Nov 21 20:09:01] NOTICE[9069][C-0001689e]: chan_sip.c:29713
sip_request_call: Conflicting extension values given. Using
'823' and not '01342244560'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/823*:5@78.11.22.33/01342244560
[Nov 21 20:09:01] NOTICE[12287][C-0001689e]: chan_sip.c:22914
handle_response_invite: Failed to authenticate on INVITE to 'Leandro
Dardini sip:100@91.11.22.33;tag=as1c0d8470'
-- SIP/78.11.22.33-000144c3 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Which is the correct syntax to use to dial directly with username and
password?

Leandro
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Re: [asterisk-users] SIP Presence across two servers

[Leandro Dardini]

Aligning presence over multiple servers is not simple and require some
changes on the dialplan and some custom code to transmit the state from one
server to the other.

The BLF on the phone is displayed using the hint of an extension. To be
able to manually manage the hint of an extension, you need to first link
the internal hint to the Custom hint. In the extensions.conf just add:

exten = _.,hint,Custom:${EXTEN}

I was unable to create the same entry in the AEL language or in the
realtime extensions table... if any was able, I will appreciate.

If a phone want to know the status for the 100-TEST sip account, it will
poll the hint for 100-TEST and in the end, it will check the status for
Custom:100-TEST.

Now you need an application to capture the change in status of every
extension on server A and send it to server B, so the Custom:100-TEST will
have the same value on both servers.

I solved this problem creating a small pair of php application, using
Asterisk Manager Interface to continuously listen to events. If I see a
phone dialing out, I change its Custom state to IN_USE... if he hangups, I
change the state back to AVAILABLE ... if it is ringing, I change the state
in RINGING and so on. You need to take into account multiple calls can be
made by the same phone and so it is not really so straightforward. When the
php AMI application identify a change in the state for a phone, it notifies
the same application running on the other server about the change, so both
asterisk are taken aligned.

Let me know if you need additional details.

Leandro



2013/11/13 Lincoln King-Cliby linc...@controlworks.com

 Hi All,



 We’ve been running Asterisk for years in our offices but just recently
 replaced an Asterisk Appliance* in our smaller office with an actual
 server, upgraded the server in hardware in our HQ location and upgrading
 both ends to 11.5.0 with Gareth’s patch for Cisco phones.

 99.99% of our endpoints are Cisco 7961Gs.



 Each office is more-or-less standalone for ease of management and fault
 tolerance but we have a unified dialplan and SIP “trunking” from site to
 site via our VPN.



 Everything presence related works wonderfully for local users, but I’m
 hoping there’s a way we could get presence for the people “at the other end
 of the pipe” fairly transparently.

 We have a lot of cross-office collaboration, and our office
 manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF)
 would love to “at a glance” know if the remote folks are available for a
 call or not.



 I’m sure this has been covered, but my Googlefu us turning up a ton of
 redundant, old, and deprecated information so I’ve resorted to asking here.

 From what I have found it sounds like it may be “easier” with IAX2 but my
 experiments with IAX2 haven’t yielded wonderful results and management
 prefers “SIP everywhere”



 If anyone has any pointers I’d greatly appreciate it – thanks in advance!



 Lincoln



 *- One of the worst IT decisions I’ve made for better or worse. Looked
 good on paper; in practice not a good idea for anything beyond a very
 simple SOHO.

 --

 Lincoln King-Cliby, CTS, DMC-D, CCMP-S

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



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Re: [asterisk-users] SIP Presence across two servers

[Leandro Dardini]

It seems very good! I am going to test it when I have a bit of time!

Leandro


2013/11/14 Ryan Wagoner rswago...@gmail.com

 I haven't tried it, but the res_corosync module states it will sync device
 state across servers.

 https://wiki.asterisk.org/wiki/display/AST/Corosync


 On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.comwrote:

 Aligning presence over multiple servers is not simple and require some
 changes on the dialplan and some custom code to transmit the state from one
 server to the other.

 The BLF on the phone is displayed using the hint of an extension. To be
 able to manually manage the hint of an extension, you need to first link
 the internal hint to the Custom hint. In the extensions.conf just add:

 exten = _.,hint,Custom:${EXTEN}

 I was unable to create the same entry in the AEL language or in the
 realtime extensions table... if any was able, I will appreciate.

 If a phone want to know the status for the 100-TEST sip account, it will
 poll the hint for 100-TEST and in the end, it will check the status for
 Custom:100-TEST.

 Now you need an application to capture the change in status of every
 extension on server A and send it to server B, so the Custom:100-TEST will
 have the same value on both servers.

 I solved this problem creating a small pair of php application, using
 Asterisk Manager Interface to continuously listen to events. If I see a
 phone dialing out, I change its Custom state to IN_USE... if he hangups, I
 change the state back to AVAILABLE ... if it is ringing, I change the state
 in RINGING and so on. You need to take into account multiple calls can be
 made by the same phone and so it is not really so straightforward. When the
 php AMI application identify a change in the state for a phone, it notifies
 the same application running on the other server about the change, so both
 asterisk are taken aligned.

 Let me know if you need additional details.

 Leandro



 2013/11/13 Lincoln King-Cliby linc...@controlworks.com

 Hi All,



 We’ve been running Asterisk for years in our offices but just recently
 replaced an Asterisk Appliance* in our smaller office with an actual
 server, upgraded the server in hardware in our HQ location and upgrading
 both ends to 11.5.0 with Gareth’s patch for Cisco phones.

 99.99% of our endpoints are Cisco 7961Gs.



 Each office is more-or-less standalone for ease of management and fault
 tolerance but we have a unified dialplan and SIP “trunking” from site to
 site via our VPN.



 Everything presence related works wonderfully for local users, but I’m
 hoping there’s a way we could get presence for the people “at the other end
 of the pipe” fairly transparently.

 We have a lot of cross-office collaboration, and our office
 manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF)
 would love to “at a glance” know if the remote folks are available for a
 call or not.



 I’m sure this has been covered, but my Googlefu us turning up a ton of
 redundant, old, and deprecated information so I’ve resorted to asking here.

 From what I have found it sounds like it may be “easier” with IAX2 but
 my experiments with IAX2 haven’t yielded wonderful results and management
 prefers “SIP everywhere”



 If anyone has any pointers I’d greatly appreciate it – thanks in
 advance!



 Lincoln



 *- One of the worst IT decisions I’ve made for better or worse. Looked
 good on paper; in practice not a good idea for anything beyond a very
 simple SOHO.

 --

 Lincoln King-Cliby, CTS, DMC-D, CCMP-S

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



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[asterisk-users] Queue linear unordered feature when using realtime

[Leandro Dardini]

Hello,
I was trying to use a queue in linear order and to provide the exact order
of members to dial by adjusting the uniqueid value. Obviously it doesn't
work and it seems an old problem:

https://issues.asterisk.org/jira/browse/ASTERISK-18480

Realtime configuration can't identify orders in the list of results, so
the members for the queue are returned in random order.

Anyone experiencing the same problem? How do you solve it?

Leandro
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Re: [asterisk-users] Asterisk Realtime Static Voicemail

[Leandro Dardini]

2013/11/11 John T. Bittner j...@xaccel.net

  Guys,



 I need you help on this one.



 Don’t know when this broke but we have a custom gui that runs on top of
 Asterisk running a real-time static for configurations.

 Nothing has changed with the database other than upgrades of Asterisk 10.



 Customer complained that there password was not changing when they called
 into voicemail and changed it.

 Database is running standard ast_config with the following fields.



 ++--+--+-+-++

 | Field  | Type | Null | Key | Default | Extra  |

 ++--+--+-+-++

 | id | int(11)  | NO   | PRI | NULL| auto_increment |

 | cat_metric | int(11)  | NO   | | 0   ||

 | var_metric | int(11)  | NO   | | 0   ||

 | commented  | int(11)  | NO   | | 0   ||

 | filename   | varchar(128) | NO   | | ||

 | category   | varchar(128) | NO   | | default ||

 | var_name   | varchar(128) | NO   | | ||

 | var_val| varchar(255) | NO   | | ||

 ++--+--+-+-++

 8 rows in set (0.00 sec)



 Did some tests and asterisk does change the password but in the
 /etc/asterisk/voicemail.conf file.

 Rename the file to see if it will then try the database. It recreates the
 file and changes the password.

 The issue is when it reads the password it looks at ast_config so it never
 really changes.

 Ran debug and no errors, I don’t even see it trying to update mysql



 Any idea what this could be.  The file below is an exact match of what’s
 in ast_config.



 /etc/asterisk/voicemail.conf

 ;! Automatically generated configuration file

 ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)

 ;! Generator: AppVoicemail

 ;! Creation Date: Mon Nov 11 01:12:51 2013

 ;!

 [default]

 9105 = 1234,Genee Jacobs,,,tz=|attach=|saycid=|hidefromdir=

 201 = ,Anne Long,,,tz=|attach=|saycid=|hidefromdir=|delete=

 [zonemessages]

 pacific = US/Pacific|'vm-received' Q 'digits/at' IMp

 eastern = America/New_York|'vm-received' Q 'digits/at' IMp

 central = America/Chicago|'vm-received' Q 'digits/at' IMp

 central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'

 military = Zulu|'vm-received' q 'digits/at' H N 'hours'
 'phonetic/z_p'

 gmt = Europe/London|'vm-received' q 'digits/at' H N 'hours'

 cet = Europe/Zurich|'vm-received' q 'digits/at' H N 'hours'

 hkg = Asia/Hong_Kong|'vm-received' q 'digits/at' H N 'hours'

 [general]

 format = wav49|gsm|wav

 serveremail = nwvoicem...@randrealty.com

 attach = yes

 emaildateformat = %A, %B %d, %Y at %r

 maxlogins = 3

 sendvoicemail = yes

 operator = yes

 pagerdateformat = %A, %B %d, %Y at %r

 externnotify = /usr/local/sigman/scripts/voicemailapp





 John Bittner

 CTO

  380 US Highway 46, Suite 500

 Totowa, NJ 07512

 Phone: 201.806.2602 x2405

 Fax:   201.806.2604

 Cell:   973.390.1090

 www.xaccel.net






 *CONFIDENTIALITY NOTICE: This e-mail message, including any attachments,
 is for the sole use of the intended recipient(s) and may contain
 confidential and privileged information which should not be shared or
 forwarded. Any unauthorized review, use, disclosure or distribution is
 prohibited. If you are not the intended recipient, please contact the
 sender by reply e-mail and destroy all copies of the e-mail.*



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Do you have compiled asterisk by yourself? In the Voicemail Build Option,
what option have you selected? I think you need to select ODBC Storage
and then configure ODBC on the system to connect to your database.

Leandro
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[asterisk-users] Disable the Connected Line info

[Leandro Dardini]

When you set sendrpid=yes in sip.conf, a very nice feature is activated.
When dialing an extension, the callerid of the dialed extension is returned
back on the display of the calling phone. So if you call extension 100, you
can see you are calling Ann (for example).

I want to selectively disable the transmission of this information back to
the caller. How can I do it?

I tried setting

Set(CONNECTEDLINE(num-pres)=prohib);

but it doesn't seem to sort any effect.

Where am I wrong?

Leandro
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[asterisk-users] Sending 603 Declined message

[Leandro Dardini]

In my dialplan I'd like to send a 603 Declined message to the user
placing the call. I see the commands for the Busy and Congestion, but not
the one for the Declined. Any help?

Leandro
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Re: [asterisk-users] Loopback question

[Leandro Dardini]

Is the echo application suitable to you?

Leandro


2013/5/20 CDR vene...@gmail.com

 Dear friends
 I need to loopback the audio on my channel. Did anybody on the development
 team thought about a function or app that would do that? If it is not
 clear, I mean that whatever audio I get, I send back.
 Philip

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Re: [asterisk-users] Secure Calling

[Leandro Dardini]

I think it can be worth checking the authenticate function.

http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate


2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com

  How do I make a user dial a passcode to make  calls through asterisk?

 We would like to place a phone at a client’s location for our employee but
 are afraid it may get abused by the other workers.



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Re: [asterisk-users] Passcode

[Leandro Dardini]

Again, the authenticate function can help you

Leandro


2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com

  How do I make a user dial a passcode if he wants to make an
 international call?



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Re: [asterisk-users] Dynamic realtime + queues

[Leandro Dardini]

You need a name column. This is my queue table:

CREATE TABLE IF NOT EXISTS `queue` (
  `name` varchar(128) NOT NULL,
  `musiconhold` varchar(128) DEFAULT NULL,
  `announce` varchar(128) DEFAULT NULL,
  `context` varchar(128) DEFAULT NULL,
  `timeout` int(11) DEFAULT NULL,
  `monitor_join` tinyint(1) DEFAULT NULL,
  `monitor_format` varchar(128) DEFAULT NULL,
  `queue_youarenext` varchar(128) DEFAULT NULL,
  `queue_thereare` varchar(128) DEFAULT NULL,
  `queue_callswaiting` varchar(128) DEFAULT NULL,
  `queue_holdtime` varchar(128) DEFAULT NULL,
  `queue_minutes` varchar(128) DEFAULT NULL,
  `queue_seconds` varchar(128) DEFAULT NULL,
  `queue_lessthan` varchar(128) DEFAULT NULL,
  `queue_thankyou` varchar(128) DEFAULT NULL,
  `queue_reporthold` varchar(128) DEFAULT NULL,
  `announce_frequency` int(11) DEFAULT NULL,
  `announce_round_seconds` int(11) DEFAULT NULL,
  `announce_holdtime` varchar(128) DEFAULT NULL,
  `retry` int(11) DEFAULT NULL,
  `wrapuptime` int(11) DEFAULT NULL,
  `maxlen` int(11) DEFAULT NULL,
  `servicelevel` int(11) DEFAULT NULL,
  `strategy` varchar(128) DEFAULT NULL,
  `joinempty` varchar(128) DEFAULT NULL,
  `leavewhenempty` varchar(128) DEFAULT NULL,
  `eventmemberstatus` tinyint(1) DEFAULT NULL,
  `eventwhencalled` tinyint(1) DEFAULT NULL,
  `reportholdtime` tinyint(1) DEFAULT NULL,
  `memberdelay` int(11) DEFAULT NULL,
  `weight` int(11) DEFAULT NULL,
  `timeoutrestart` tinyint(1) DEFAULT NULL,
  `periodic_announce` varchar(50) DEFAULT NULL,
  `periodic_announce_frequency` int(11) DEFAULT NULL,
  `ringinuse` tinyint(1) DEFAULT NULL,
  `setinterfacevar` tinyint(1) DEFAULT NULL,
  PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;



2013/4/18 Tommy Cooper tomcoope...@yahoo.com

 Hi,**
  
 I am trying to store queues.conf to a MySQL database using dynamic
 realtime. I have a working ODBC connection and the queueing system already
 works but I want to store the queues.conf file to a database. I am
 following the guide from Asterisk the definitive guide, the ebook can be
 found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html **
 **
 ** **
 I have a database called asterisk which contains 2 main tables: Queues and 
 queue_member_table,
 both tables have sample data.
 ** **
 mysql select * from queue_member_table;
 +--+++---+-++
 | uniqueid | membername | queue_name | interface | penalty | paused |
 +--+++---+-++
 |1 | SIP/1000   | support| SIP/1000  |NULL |   0 |
 
 +--+++---+-++
 ** **
 ** **
 SQL select QueueID,name,strategy from Queues;
 ** **
 |QueueID|  namestrategy  *
 ***
  1 support rrmemory
 
 ** **
 There are more fields but these are the most important
 ** **
 I keep getting this error:
 ** **
 node1*CLI queue show 
 No queues.
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:645
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000:
 [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL
 syntax; check the manual that corresponds to your MySQL server version for
 the right syntax to use near ''\' ORDER BY name' at line 1 (202)
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:657
 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
 asterisk [asterisk-connector]...
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:761 ast_odbc_sanity_check:
 Connection is down attempting to reconnect...
 [Apr 18 22:41:06] NOTICE[18599]: res_odbc.c:1527 odbc_obj_connect:
 Connecting asterisk
 [Apr 18 22:41:06] NOTICE[18599]: res_odbc.c:1559 odbc_obj_connect:
 res_odbc: Connected to asterisk [asterisk-connector]
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:645
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000:
 [MySQL][ODBC 5.1 Driver][mysqld-5.1.67]You have an error in your SQL
 syntax; check the manual that corresponds to your MySQL server version for
 the right syntax to use near ''\' ORDER BY name' at line 1 (202)
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:657
 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
 asterisk [asterisk-connector]...
 [Apr 18 22:41:06] WARNING[18599]: res_odbc.c:761 ast_odbc_sanity_check:
 Connection is down attempting to reconnect...
 [Apr 18 22:41:06] NOTICE[18599]: res_odbc.c:1527 odbc_obj_connect:
 Connecting asterisk
 [Apr 18 22:41:06] NOTICE[18599]: res_odbc.c:1559 odbc_obj_connect:
 res_odbc: Connected to asterisk [asterisk-connector]
  
 ** **
 ** **
 extensions.conf:   
 ** **
 ** **
   
 [general]
 autofallthrough=yes
 ** **
 [default]
 exten = 1000,1, Dial (SIP/1000)

Re: [asterisk-users] Dynamic realtime + queues

[Leandro Dardini]

Uhm ... I see the easy way will be to tcpdump the connection between the
asterisk and the mysql database server and to dump the exact SQL syntax
used. It will be something wrong...

Leandro

PS
tcpdump -i any -n -s 1500 -w /tmp/data.pcap port 3306




2013/4/18 Tommy Cooper tomcoope...@yahoo.com

 Thank you for your response

 I already have a name column but my primary key is 'QueueID' instead of
 name


 +-+---+--+-+++
 | Field   | Type  | Null | Key |
 Default| Extra  |

 +-+---+--+-+++
 | QueueID | mediumint(8) unsigned | NO   | PRI |
 NULL   | auto_increment |
 | name| varchar(128)  | NO   | UNI |
 NULL   ||
 | description | varchar(128)  | YES  | |
 NULL   ||
 | maxlen  | tinyint(4)| YES  | |
 NULL   ||
 | reportholdtime  | varchar(3)| YES  | |
 no ||
 | periodic_announce_frequency | varchar(4)| YES  | |
 NULL   ||
 | periodic_announce   | varchar(128)  | YES  | |
 NULL   ||
 | strategy| varchar(20)   | NO   | |
 rrmemory   ||
 | joinempty   | varchar(35)   | YES  | |
 no ||
 | leavewhenempty  | varchar(35)   | YES  | |
 no ||
 | autopause   | varchar(3)| YES  | |
 no ||
 | announce_round_seconds  | varchar(4)| YES  | |
 NULL   ||
 | retry   | varchar(4)| YES  | |
 NULL   ||
 | wrapuptime  | varchar(4)| YES  | |
 NULL   ||
 | announce_holdtime   | varchar(3)| YES  | |
 no ||
 | announce_frequency  | varchar(4)| YES  | |
 0  ||
 | timeout | varchar(4)| YES  | |
 60 ||
 | context | varchar(128)  | NO   | |
 NULL   ||
 | musicclass  | varchar(128)  | YES  | |
 default||
 | autofill| varchar(3)| YES  | |
 yes||
 | ringinuse   | varchar(45)   | YES  | |
 no ||
 | musiconhold | varchar(128)  | YES  | |
 yes||
 | monitor_type| varchar(128)  | YES  | |
 MixMonitor ||
 | monitor_format  | varchar(128)  | YES  | |
 wav||
 | servicelevel| varchar(4)| YES  | |
 60 ||
 | queue_thankyou  | varchar(128)  | YES  |
 |||
 | queue_youarenext| varchar(128)  | YES  |
 |||
 | queue_thereare  | varchar(128)  | YES  |
 |||
 | queue_callswaiting  | varchar(128)  | YES  |
 |||
 | queue_holdtime  | varchar(128)  | YES  |
 |||
 | queue_minutes   | varchar(128)  | YES  |
 |||
 | queue_seconds   | varchar(128)  | YES  |
 |||
 | queue_lessthan  | varchar(128)  | YES  |
 |||
 | queue_reporthold| varchar(128)  | YES  |
 |||
 | relative_periodic_announce  | varchar(4)| YES  | |
 yes||

 +-+---+--+-+++
 35 rows in set (0.00 sec)

  - Forwarded Message -
 *From:* Leandro Dardini ldard...@gmail.com
 *To:* Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List -
 Non-Commercial Discussion asterisk-users@lists.digium.com
 *Sent:* Thursday, April 18, 2013 11:04 PM
 *Subject:* Re: [asterisk-users] Dynamic realtime + queues

  You need a name column. This is my queue table:

  CREATE TABLE IF NOT EXISTS `queue` (
   `name` varchar(128) NOT NULL,
   `musiconhold` varchar(128) DEFAULT NULL,
   `announce` varchar(128) DEFAULT NULL,
   `context` varchar(128) DEFAULT NULL,
   `timeout` int(11) DEFAULT NULL,
   `monitor_join` tinyint(1) DEFAULT NULL,
   `monitor_format

[asterisk-users] rtcachefriends and rtautoclear on change password

[Leandro Dardini]

Hello friends,
I am using from a long time rtcachefirends=yes and rtautoclear=yes in
my sip.conf for asterisk 11.2.1.

I have found the data of the peers are never reloaded from the
database, so if you change the password for a peer, it will continue
to work with the old password. Do you think it is the expected
behaviour?

From the documentation for rtautoclear=yes

If set to yes, when the registration expires, the friend will
vanish from the configuration until requested again. If set
to an integer, friends expire within this number of seconds
instead of the registration interval.

The phone will renew the registration before it expires, so maybe it
never expires.

I have tried to set the rtautoclear to 60, but the result is the same,
the new password is never enforced.

Any suggestion apart from removing the rtcachefriends?

Leandro

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Re: [asterisk-users] rtcachefriends and rtautoclear on change password

[Leandro Dardini]

You are right for the commands to prune and clear the cache. But what is
the meaning of the meaning of the configuration parameter rtautoclear if it
is not clearing the cache?

Leandro

I am typing from my mobile phone...
Il giorno 26/mar/2013 14:38, Michael L. Young myo...@acsacc.com ha
scritto:

 - Original Message -
  From: Leandro Dardini ldard...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
  Sent: Tuesday, March 26, 2013 5:28:22 AM
  Subject: [asterisk-users] rtcachefriends and rtautoclear on change
 password
 
  Hello friends,
  I am using from a long time rtcachefirends=yes and rtautoclear=yes in
  my sip.conf for asterisk 11.2.1.
 
  I have found the data of the peers are never reloaded from the
  database, so if you change the password for a peer, it will continue
  to work with the old password. Do you think it is the expected
  behaviour?
 
  From the documentation for rtautoclear=yes
 
  If set to yes, when the registration expires, the friend will
  vanish from the configuration until requested again. If set
  to an integer, friends expire within this number of seconds
  instead of the registration interval.
 
  The phone will renew the registration before it expires, so maybe it
  never expires.
 
  I have tried to set the rtautoclear to 60, but the result is the
  same,
  the new password is never enforced.
 
  Any suggestion apart from removing the rtcachefriends?

 With rtcachefriends turned on, the realtime peer is cached in memory.
  Therefore, in order to clear the cache for that peer, you should check
 into issuing the command sip prune realtime peer peername if you want
 to clear out only the one peer.  If you want to reload the peer back in
 memory after clearing it out, you can issue sip show peer peername load
 to load it back from the db.

 Michael

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Re: [asterisk-users] Optimizing Asterisk Environment

[Leandro Dardini]

I dont apply any secret recipe while installing asterisk, but maybe you can
share yours...

I am typing from my mobile phone...
Il giorno 23/mar/2013 14:34, Nick Khamis sym...@gmail.com ha scritto:

 Hello Everyone,

 We are getting some rather poor results (relative) with our Asterisk
 setup. Not sure if we are using the sipp correctly etc.. but
 nevertheless, is there any documentation that describes how we can get
 the most our of our Asterisk box. For example when we hit the too
 many file error, and fixing it using ulimit. Also, is there any
 way we can allocate sufficient memory to our Asterisk instance when
 starting the PBX.

 An up to date and in-depth tutorial that covers this would be great. A
 quick search yielded pretty motivating success stories, but no little
 to no description on how to achieve them.


 Kind Regards,

 Nick.

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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[Leandro Dardini]

2013/3/21 Florian Wolters flor...@florian-wolters.de:
 Hi @ll,

 I just moved my Asterisk Box and changed the Provider and Internet Access to 
 a full IP Access by Deutsche Telekom.

 I set up my sip.conf as I found various examples throughout the Net. Calls 
 and some other stuff is basically working.

 The problem I ran into is, that the outgoing and incoming calls are dropped 
 after exactly 15 Minutes. Solution for this should be setting the 
 session-timers to refuse but this doesnt change anything here.

 I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest 
 Asterisk by Digium without success.

 Has anyone else has the Same problem or is a solution already known? Could 
 someone point me in the right direction? I can provide (debug) logs if 
 essential.

 Best regards

Flo



I think it is important to know the reason the call is disconnected.
Start checking who is sending the BYE and if before the BYE there is
other weird packets, like retry of packet sending ...

A simple tcpdump can help explain all the mistery.

Leandro

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Re: [asterisk-users] Need help understanding CDR

[Leandro Dardini]

Top replying ...

In the CDR you have two fields, duration and billed. Duration is
the total time from Dial command to end of calls. It is the time the
Dial command is running. Billed is the time from when the other
party answered and the end of the call.

In your example, duration and billsec will differ for just a second,
the time from the Call Connected to asterisk and the Welcome
greeting starts.

Leandro

2013/3/18 RSCL Mumbai rscl.mum...@gmail.com:
 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[Leandro Dardini]

You can add custom fields in the CDR, so your dialplan can store start
time, end time and duration whenever you like.

Just use something like the

Set(CDR(customfield)=100);

Leandro

2013/3/18 RSCL Mumbai rscl.mum...@gmail.com:
 Thank you every one.
 Now I understand why I was confused.
 I have always been using Asterisk in an Inbound environment.
 Hence my thought were misaligned wrt answered  billed.
 Now I understand. Thank you all!!

 Is there anyway to capture the time for conversation, IVR, hold etc etc.
 If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any 3rd
 party application, more suitable for an Inbound environment.

 It would help a lot if I could capture fragmented distribution of time per
 call -- time in IVR, Queue, Call etc.

 Regards,
 Sans









 On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 hi,

 00:00 -- Call Connected to asterisk - duration start here
 00:01 -- welcome greeting starts  billisec start here

 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
 --- both end here

 duration = 01:15
 bilsec = 01:14

 duration start as soon as call arrived in asterisk.
 bilsec start as soon as call answered.

 exten s,1,Answer()  duration and bilsec start at same time
 because you answered the call immidataly
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec are same

 exten s,1,Ringing(10) -- duration start here
 exten s,n,Answer()  bilsec start here
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec end here

 so billsec is 10 seconds less then duration

 hope this will help you.

 On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.com
 wrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed
 time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com
 wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2
 sec correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] asterisk sizing for play and dtmf detection

[Leandro Dardini]

2013/3/8 nik600 nik...@gmail.com

 Dear all

 i'm planning a migration to asterisk for a high volume IVR service
 (from 1000 to 1500 concurrent call)

 The IVR service is based only on DTMF tones so the features required is

 - play feature
 - dtmf detection

 Asterisk will receive calls via VOIP (SIP with g711 codec)

 The IVR service wil be a static service based on Asterisk dialplan
 with some prompt (from 0 to 5, play of files in the same codec of the
 received call) and some dtmf detections.

 How many simultaneous call can i handle per server? each server will have:

 4 core 3.0 Ghz
 4 GB of RAM

 I need an aproximate sizing:

 0-100 calls per server ?
 100-200 calls per server ?
 200-300 calls per server ?
 300-400 calls per server?
 400-500 calls per server?

 Thanks to all in advance

 --
 /*/
 nik600
 http://www.kumbe.it

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The bigger server I have has 150 max channels during peak hours and has a
max load of 0.5 with 24 cores. When I was using a 4 cores server with the
same number of channels, I get a load of 3 ... so the load x core relation
is valid. I think it will be good to have a load not over 4 for a 4 core
server, so you can have at least 200 active channels on the server. If you
accept more load, then you can get more channels.

Leandro
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Re: [asterisk-users] asterisk with 1000 extensions

[Leandro Dardini]

2013/3/7 Steve Edwards asterisk@sedwards.com

 Please don't top-post.


 On Thu, 7 Mar 2013, Bharat Lalcheta wrote:

  You can use ATA box with pstn phone to reduce cost.


 Are you wiring a building where multiple-line SIP gateways make sense?

 How about a description of what you are trying to do?

 Personally, I like Polycom SIP phones but I don't have to buy 1,000 of
 them :)



I bet it is a school assignment ... home work or the way you like to call
them. However I have a box with 972 peers, no reinvite (but no
transcoding), average usage of conference call and other audio mix feature,
reaching a max of 60 CPS and an average of 150 channels without problems.
The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works
fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150  @ 2.66GHz

Leandro
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Re: [asterisk-users] asterisk with 1000 extensions

[Leandro Dardini]

2013/3/7 Duncan Turnbull dun...@e-simple.co.nz


 On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:


 On Thu, 7 Mar 2013, Bharat Lalcheta wrote:

 You can use ATA box with pstn phone to reduce cost.


 Are you wiring a building where multiple-line SIP gateways make sense?

 How about a description of what you are trying to do?

 Personally, I like Polycom SIP phones but I don't have to buy 1,000 of
 them :)



 This is not school assignment or home work :)  We need to setup in society
 buildings. Each flat will have SIP extension (hard phone) registered on
 asterisk server. Calling between SIP extensions is required. No PSTN /
 ITSP SIP trunking. Just like inter-com feature.

 One way is to install 1000 IP Phones one at each flat
 Secondly, install multiple-line SIP gateways with RJ-11 cabling.

 Is there any other low budget solution for this setup?

 Your costs will be in the handsets. Yealink make good cheap phones, you
 need to find a supplier who can do you a great deal on 1000 phones

 http://www.yealink.com/product_info.aspx?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147

 But I am not sure why you cant use analogue phones and SIP channel banks
 such as grandstream or USB ones such as Xorcom. The per line cost will come
 down and you only need telephony grade cabling to the premise. You can get
 $10 phones which limit the desire of people to walk off with them

 The server and setup will cost nothing compared to the handsets



 Thanks,
 Kamlesh

 Sorry, but it is not the first time we help little boys to make homework,
it seems asterisk course are common in India and it is easier to cheat than
to apply.

If you are really trying to serve 1000 phones, beside the usage of SIP or
analogue phones via channel banks, I think it will be better to not handle
all the load on a single server, but to spread the phone among multiple
servers. The best will be to have multiple asterisks working together using
realtime extensions. It is not difficult to make.

Leandro
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Re: [asterisk-users] Extension cant pickup calls but can transfer.

[Leandro Dardini]

If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.

Leandro

I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, Luis H. Forchesatto 
luisforchesa...@gmail.com ha scritto:

 Greetings.

 I got an extension on my Elastix who cannot pick calls on the other
 extensions, but It can transfer his calls to the other extensions. When
 this extension tries to pickup a call pressing *8  it simply does not pick
 it up. Transfering calls works just fine so dtmf may be not the problem.

 Where should I look?

 Any further information needed just ask.

 --
 Att.*
 ***


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Re: [asterisk-users] Delay before audio starts

[Leandro Dardini]

I think a simple tcpdump of the traffic will show the mystery. It can be
your provider doing something nasty. Have you tried using some other cheap
SIP termination? or arrange a fake termination yourself on another server?

Leandro

2013/3/1 Gerard gsara...@rarcoa.com

 I thought it was the re-invites too, but I have it turned off everywhere.

 On 03/01/13 08:36, Eric Wieling wrote:
  When Answer fixes the issue, the root cause is often NAT (could be
 firewall) since Answering the call prevents any reinvites.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
  Sent: Friday, March 01, 2013 9:33 AM
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] Delay before audio starts
 
  I've found a workaround of sorts, If I change my below code to :
   1AA = {
   NoOp(${CALLERID(num)});
 Answer();  // --- add this
   Ringing;
   Set(CHANNEL(musicclass)=none);
   Dial(${OUTBOUND-TRUNKR}/1XX,30);
   Voicemail(198,u);
};
 
  That fixes the issue. It doesn't fix the call forward issue on the phone
 though. I've made a few extra extensions, one each corresponding to a
 number he wants to call forward to, if I have him forward to the extensions
 who then forward to the real number, it works, thanks to adding Answer()
 to the dialplan.
 
  -Gerard
 
 
  On 02/26/13 13:19, Gerard wrote:
  Hi everyone,
 
  I'm having a hard time figuring this issue out, we just switched from
  a
  T1 PRI to a SIP trunk provider and that's when the issue started.
  Now when someone forwards all calls on their phone to a cellphone,
  when a customer calls in, Asterisk correctly calls the cellphone and
  connects the call, but there is a long delay before the audio starts,
  basically for the first 6-10 seconds of the call there is dead
  silence, eventually the audio will start and everything works correctly.
  We never had this problem with the PRI. So I suspect it has something
  to do with a call coming in as SIP and going out as SIP.
 
  At first I thought it was a call forwarding issue because I got this
  message in the console:
  [Feb 26 12:35:19] NOTICE[1143][C-025d]: app_dial.c:958 do_forward:
  Not accepting call completion offers from call-forward recipient
  Local/1XX@default-0013;1
 
  So I put this in my dial plan:
 
  1AA = {
  NoOp(${CALLERID(num)});
  Ringing;
  Set(CHANNEL(musicclass)=none);
  Dial(${OUTBOUND-TRUNKR}/1XX,30);
  Voicemail(198,u);
   };
 
  So basically as soon as someone calls incoming number AA,
  Asterisk dials phone number XX. it's a quick and dirty way to
  call forward.. and this does the same thing, there's a good 8 second
  delay before the audio kicks in.
 
 
  There is a Linux firewall with NAT in the path, but I have no other
  audio issues, so don't *think* it's a factor.
  I just upgraded to asterisk 11.2.1.
 
 
  Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
  2013-02-23 01:40:02 UTC
 
 
  Any help would be appreciated,
  Thanks,
 
 
 
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 (630) 654-3556 (fax)
 (630) 915-4122 (cell)

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[asterisk-users] AEL Macro are evil :-)

[Leandro Dardini]

I just discover an hidden problem with AEL macro I want to have your
feedback. If you use a macro to dial out, like dialout(${EXTEN}), the leg
extension will became s and if it happens you transfer the call,
that will be the callerid appearing on the other phone display.
I am just rewriting all the dialplan getting rid of the macro and using
gosub, even if asterisk is complaining about  application call to gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!, but I am not seeing any other way...

Leandro
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Re: [asterisk-users] AEL Macro are evil :-)

[Leandro Dardini]

Knowing the exten variable of the original dialplan is not a problem. The
problem is just when the call is transferred via blind transfer. Asterisk
does a little magic with the callerid of the legs and using Macro, just
breaks it.

If Joe at ext 100 call Sally at ext 200 and then Joe transfers the call to
Bob at ext 300, then Bob will see the callerid 200 on his phone. That is
not true if the dial is made inside a Macro. In this way, Bob will see
s

The macro can be something as simple as:

macro dialpeer(number) {
   dial(SIP/number);
}

Leandro

2013/2/24 Mitul Limbani mi...@enterux.in

 Hi,

 You might want to use ${MACRO_EXTEN} variable inside to preserve exten
 variable of the original dialplan exten variable.

 Mitul
 On Feb 24, 2013 4:04 PM, Leandro Dardini ldard...@gmail.com wrote:

 I just discover an hidden problem with AEL macro I want to have your
 feedback. If you use a macro to dial out, like dialout(${EXTEN}), the leg
 extension will became s and if it happens you transfer the call,
 that will be the callerid appearing on the other phone display.
 I am just rewriting all the dialplan getting rid of the macro and using
 gosub, even if asterisk is complaining about  application call to gosub
 affects flow of control, and needs to be re-written using AEL if, while,
 goto, etc. keywords instead!, but I am not seeing any other way...

 Leandro



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Re: [asterisk-users] Remove Abandoned call

[Leandro Dardini]

2013/2/21 akhilesh chand omakhileshch...@gmail.com

 hello all,

 i have two asterisk server for call transfer and one more asterisk server
 for agent login(server_X) where agent take the call.

 server_A  and server_B
 server_A is connected with pri and configure with 60 channel for call
 transfer into server_X
 server_B is connected with pri and configure with 30 channel for call
 transfer into server_X

 my query is that some time two call originate same time from two different
 server_A and server_B and hit into server_X and one call is abandoned and
 another one have taken by the agent
 But i don't want to abandoned the call, I want to set the priority,
 supposed to server_A and server_B call originate same time server_X take
 the call from server_A first and then take the call server_B after 1 sec

 please guide me

 Regards
 Akhilesh


I am sorry if I haven't completely understood your question, but english is
not my native language. If calls from server_A and server_B are put in the
same queue in server_X, how can one of them being abandoned? Calls will be
processed in the same order as they arrive.

Leandro
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Re: [asterisk-users] Playback on h exten

[Leandro Dardini]

2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it

 Hi all, I'm trying to setup a Quiz/feedback for caller of call center when
 a agent hangup.
 I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c
 and g but every time I try to play something I got:

 -- Executing [301@from-test:1] Dial(SIP/300-0045,
 SIP/301,60,rjtTg) in new stack
 -- Called SIP/301
 -- SIP/301-0046 is ringing
 -- SIP/301-0046 answered SIP/300-0045
 -- Auto fallthrough, channel 'SIP/300-0045' status is 'ANSWER'
 -- Executing [h@from-test:1] Goto(SIP/300-0045, play,s,1) in
 new stack
 -- Goto (play,s,1)
 -- Executing [s@play:1] NoOp(SIP/300-0045, play) in new stack
 -- Executing [s@play:2] SayDigits(SIP/300-0045, 123579) in
 new stack
 [Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback:
 Failed to write frame
 -- SIP/300-0045 Playing 'digits/1.ulaw' (language 'en')
   == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-0045'

 This is my dialplan:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 exten = h,1,Goto(play,s,1)

 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)


 Anyone can help me?

 Thanks

 Enrico.


If you choose to go with the Dial command and use the g option, you have
not to use the h extension, but just provide a next priority. Your
dialplan has to be:

[from-test]
exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
*exten = _X.,2,Goto(play,s,1)*

[play]
exten = s,1,Noop(play)
exten = s,2,Saydigits(123579)

Leandro
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Re: [asterisk-users] Playback on h exten

[Leandro Dardini]

The h exten is triggered when the channel is hangup, so you cannot send any
voice data on it.

Leandro

2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it

 Yes, correct now it works for Dial.
 I think is the same with c option on Queue, do you think there's a way
 to do it on h exten?
 My goal is to inject my dialplan on hangup macro.

 Enrico.
 --


 If you choose to go with the Dial command and use the g option, you have
 not to use the h extension, but just provide a next priority. Your
 dialplan has to be:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 *exten = _X.,2,Goto(play,s,1)*

 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)

 Leandro


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 Pasqualotto Enrico
 cell. +39 3473292620
 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto
 http://www.netspin.it :: e.pasqualo...@netspin.it

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Re: [asterisk-users] Asterisk question

[Leandro Dardini]

2013/2/20 Nguyễn Công nguyencong.1...@gmail.com

 Hello everyone, I’m new to Asterisk and I have a question. There is a
 phone call between two users, then they are talking to each other directly
 or by the server. I mean all packets from the user A to user B will be send
 directly to each other or will those packets from user A must be send to
 server and server will send to user B.

 Thanks.

 --


Both cases can happens. In a VoIP call we have two connections, one is used
for signaling, usually port 5060 for SIP protocol, UDP transport and one is
used for media (voice), usually random port. When the call starts the
asterisk server sits in the middle of the media path, meaning all voice
packets from phone A go to asterisk server and they are rerouted to phone
B. After few milliseconds, if configured this way, asterisk server
instructs the phone A to send the media directly to phone B to save
bandwidth. It is named reinvite

Leandro
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Re: [asterisk-users] problem to socket programming in AGI

[Leandro Dardini]

Check if you have selinux enforcing anf try to disable it

I am typing from my mobile phone...
Il giorno 04/feb/2013 18:43, C. Savinovich c.savinov...@itntelecom.com
ha scritto:


 I would just type in the web service url manually in a browser, and if the
 browser displays the response, then there it is, the connection to the host
 server is open.

 Christian Savinovich
 *VoIP  Telephony Consultant*
 646-982-3572



   Original Message 
 Subject: Re: [asterisk-users] problem to socket programming in AGI
 From: Justin Killen jkil...@allamericanasphalt.com
 Date: Mon, February 04, 2013 12:25 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com

  Yes, I think curl would probably be a better option than trying to use
 sockets directly, but if the socket won’t connect it doesn’t really matter
 what higher level method is used. 
  -Justin 
   --
  *From:* asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
 *On Behalf Of *C. Savinovich
 *Sent:* Monday, February 04, 2013 9:16 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] problem to socket programming in AGI
  ** **
  ** **
  I don't get it, if it is a web service, why do you use sockets? Isn't it
 just a matter of calling the web service using curl,and then wait for the
 response? what am I missing?
  ** **
  Christian Savinovich
  *VoIP  Telephony Consultant*
  646-982-3572
   
  ** **

   Original Message 
 Subject: Re: [asterisk-users] problem to socket programming in AGI
 From: Justin Killen jkil...@allamericanasphalt.com
 Date: Mon, February 04, 2013 12:05 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com


 
 **
  You are correct, this is not an asterisk question.  What I would suggest
 would be to run your script outside of asterisk and debug the connection.
 Looking at the php doc page for fsockopen (*
 http://php.net/manual/en/function.fsockopen.php*http://php.net/manual/en/function.fsockopen.php),
 I see this example:
  ?php
 $fp = fsockopen(www.example.com, 80, $errno, $errstr, 30);
 if (!$fp) {
 echo $errstr ($errno)br /\n;
 } else {
 $out = GET / HTTP/1.1\r\n;
 $out .= Host: *www.example.com* http://www.example.com\r\n;
 $out .= Connection: Close\r\n\r\n;
 fwrite($fp, $out);
 while (!feof($fp)) {
 echo fgets($fp, 128);
 }
 fclose($fp);
 }
 ? 
  
 ** **
  I would first try running that (put in your host and port) and see what
 the error string coming back is.
  ** **
   -Justin
--
  *From:* 
 *asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com[
 *mailto:asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com]
 *On Behalf Of *Muhammad
 *Sent:* Monday, February 04, 2013 5:07 AM
 *To:* **Asterisk Users Mailing List - Non-Commercial Discussion**
 *Subject:* [asterisk-users] problem to socket programming in AGI
 
  ** **
   Hi,
 I know maybe this question is not related to asterisk, but I want to make
 XML RPC web service to other http server.
 I have elastix system. it is https and problem is source not destination
 server. In xml rpc we have fsockopen connection to connect destination
 server(xml rpc server). It return me connect error(0).

 what is the problem. is this related to elastix(asterisk) server?
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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

[Leandro Dardini]

2013/1/31 Ishfaq Malik i...@pack-net.co.uk

 On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
  On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:
  Thanks - I was hoping there was some silver bullet to use out
  there. Thanks
  anyway.
 
 
  There is.  If you build a reliable network, the phones will simply
  never have a problem.  We've got customers with phones that have never
  lost contact for years.  Re-registering is just a crutch for a network
  defect.
 
 
  --
  Carlos Alvarez
  TelEvolve
  602-889-3003
 
 
 This is so true!


If you have no NAT or dynamic IP in your network, you can just remove the
registration process and assign to each peer its IP address.

Leandro
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Re: [asterisk-users] Asterisk Messaging Refuses To Work!

[Leandro Dardini]

2013/1/30 XBrian bobo...@yahoo.co.uk

 I am pulling my hairs out here. This is my dialplan.

 exten = 100,1,Set(AGISIGHUP=no)
 exten = 100,n,AGI(a2billing.php,4,callingcard)
 exten = 100,n,Set(__APP_MSG_IND=${APP_MSG_IND})
 exten = 100,n,Set(__APP_MESSAGE=${APP_MESSAGE})
 exten = 100,n,Hangup()

 exten = h,1,GotoIf($[${APP_MSG_IND} = YES]?send-msg,1)
 exten = h,n,Hangup()

 exten = send-msg,1,SendText(${APP_MESSAGE})
 exten = send-msg,n,Hangup()

 I can see on the command line that the SendText() is actually being
 called, but
 the softphone isnt getting the text.
 What am I doing wrong?
 Is there a variable to be set?

 Any ideas will be most welcome



If I was in your shoes (is this the right English sentence?)  I'll run a
tcpdump command to check the content of the SIP packet containing the
message. That way you'll know if the asterisk or the softphone is to blame.

Leandro
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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

[Leandro Dardini]

You can just shorten the time the phone device register on the asterisk
server. It is up to the peer to send the registration command. It cannot be
triggered or forced in any way.

Leandro

2013/1/30 XBrian bobo...@yahoo.co.uk

 I am aware that the direction is from peer to asterisk.  Its
 a valid question. If a solution did exist, guarantees near 100 per cent
 availability. Especially if the device is actually there.





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Re: [asterisk-users] round-robin in asterisk 1.4

[Leandro Dardini]

The simplest way is to use the Random function and to pickup one number
from 1 to 3 and use that line.

Leandro

I am typing from my mobile phone...
Il giorno 29/gen/2013 11:35, Salaheddine Elharit 
salah.elharit...@gmail.com ha scritto:

 I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210
 with 2 port E1.

 now i bought another card Diguim TE410 and I want to add it

 the current configuration : connection (WIMAX) from the first ISP and
 connection (fiber optic) from the secend ISP.

 the desired configuration : connection (WIMAX) and connection (radio beam)
 from the first ISP.from the second ISP no change (still have the fibre
 optic)

 my question how to active the round-robin in asterisk 1.4 in order to
 active the 3 technology (WIMAX-radio beam and fibre optic)
 any help please

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Re: [asterisk-users] Complex Call Distribution

[Leandro Dardini]

2013/1/26 RSCL Mumbai rscl.mum...@gmail.com

 Hello,

 I have Elastix ISO install (FreePBX 2.7.0.3)

 My current Setup is as follows:
 Inbound Route  Queue  (Dynamic Agents)

 The queue distributes calls based on rrMemory.

 I have been asked to redesign the call distribution as follows:

 Calls will be delievered to Level-1 Agents (say 4 dynamic agents) in
 rrMemory format.
 When Level-1 Agents are busy, distribute calls to Level-2 Agents (say 3
 dynamic agents) in rrMemory format.
 When Level-2 Agents are busy, distribute calls to Level-3 Agents (say 2
 dynamic agents) in rrMemory format.

 Is it possible to setup the call distribution in the above format using
 any kind of logic or algorithm ?

 I tried using Penalties function in Queues.
 Created 2 penalties : 0 (level-1) and 1000 (level-2) and assigned
 penalties to agents (static)
 I made a few test calls, but Level-2 agents were delivered calls inspite
 of Level-1 agents being available.

 Any help or pointers are appreciated.

 Thx,
 Vai


I know for sure how to do it in asterisk, but I don't know how to do it
using elastix interface. Maybe you can have more luck asking to some
elastix related mailing list.

Leandro
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

It is a shame we were unable to find the solution to your problem. Do you
want to setup a test system like the good one and let me access it to check
what is going on? I am really really curious.

Leandro
Il giorno 26/gen/2013 19:49, Dan Journo d...@keshercommunications.com ha
scritto:

  It is really unbelievable ... I was thinking: Asterisk uses an internal
 database to maintain states of peers. It is usually located in
 /var/lib/asterisk/astdb and it is a berkely db, but other database backends
 seem available. Are you sharing also this database between the two servers?
 It is the only option left...

 ** **

 The only thing shared is the sip realtime db.

 ** **

 I think i'm going to try removing the sip realtime db and automate the
 creation of the sip.conf file and issuing of the 'sip reload' and see if
 the problem goes away.

 ** **

 ** **

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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/25 Dan Journo d...@keshercommunications.com

  Upgrading to the latest version didn't help. After about 30 minutes,
 Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
 Registered on Asterisk1.

  It is something really amazing... Can you run sip show peers on each
 one of the servers and post the response?

 ** **

  You said the second asterisk is completely opaque to your peers. Can
 you run a tcpdump on secondary server to see if for some obscure reason the
 phones try to contact the secondary asterisk?

 ** **

 I'll monitor one peer using tcpdump over a few hours and then review the
 packets. However, SIP DEBUG isn't showing any REGISTER packets.

 ** **

 Here's the sip peers output. Values and names have been hidden. Some
 appear as Unreachable the secondary server and some appear as OK.

 I think some are listed as OK because the endpoint routers are performing
 some type of SIP ALG and routing packets based on port number and not
 source ip address.

 However, from the SIP DEBUG output, it seems clear that the secondary
 server in this example is sending out Keepalives based on the information
 that the primary server has entered into the realtime DB.

 ** **

 *Show peers Output from a primary server*

 Name/username  HostDyn
 Forcerport ACL Port Status Realtime

 a201/A201  217.x.x.48D
 N 65229OK (88 ms) Cached RT

 a202 (Unspecified)D
 N 0UNREACHABLE Cached RT

 b201/44845287  78.x.x.101   D   N
 5060 OK (26 ms) Cached RT

 c201/s   193.x.x.174  D   N
 5060 OK (52 ms) Cached RT

 d201/d201  94.x.x.228 D   N
 5060 OK (33 ms) Cached RT

 e201/e20194.x.x.44 D   N
 55018OK (40 ms) Cached RT

 e202/e20294.x.x.44 D   N
 55022OK (46 ms) Cached RT

 e203/e20394.x.x.44 D   N
 55024OK (40 ms) Cached RT

 e204/e20494.x.x.44 D   N
 55008OK (40 ms) Cached RT

 e205/e20594.x.x.44 D   N
 55016OK (41 ms) Cached RT

 e206/e20694.x.x.44 D   N
 55014OK (40 ms) Cached RT

 e207/e20794.x.x.44 D   N
 55020OK (41 ms) Cached RT

 e208/e20894.x.x.44 D   N
 5060 OK (41 ms) Cached RT

 e209/e20994.x.x.44 D   N
 55012OK (40 ms) Cached RT

 e210/e21094.x.x.44 D   N
 55010OK (41 ms) Cached RT

 e211/e21194.x.x.44 D   N
 55026OK (38 ms) Cached RT

 e212/e21281.x.x.93D   N
 5060 OK (46 ms) Cached RT

 f201 (Unspecified)D   N
 0UNREACHABLE Cached RT

 g201/g  78.x.x.207 D   N 5060
 OK (29 ms) Cached RT

 h201/h201  217.x.x.78   D   N 38980
 OK (22 ms) Cached RT

 i201 (Unspecified)D   N
 0UNREACHABLE Cached RT

 i203/ i203 109.x.x.103  D   N 5060
 OK (32 ms) Cached RT

 i204/ i204 109.x.x.103  D   N 1025
 OK (31 ms) Cached RT

 i205/ i205 81.x.x.144 D   N
 5060 OK (32 ms) Cached RT

 i206/ i206  109.x.x.103  D   N 1035
  OK (31 ms) Cached RT

 i207/ i207  109.x.x.103  D   N
 1032 OK (32 ms) Cached RT

 i208/ i208 109.x.x.103  D   N 1024
 OK (31 ms) Cached RT

 j201/s   94.x.x.62D   N
 57813OK (35 ms) Cached RT

 o201/o201  92.x.x.86 D   N
 51824OK (47 ms) Cached RT

 o202/o202  92.x.x.86 D   N
 58641OK (48 ms) Cached RT

 o203/o203  92.x.x.86 D   N
 49172OK (47 ms) Cached RT

 j204/j204  176.x.x.214  D   N
 34824OK (49 ms) Cached RT

 k201/k201  2.x.x.169 D   N
 52757OK (53 ms) Cached RT

 k202/k202  (Unspecified)D   N
 0UNKNOWNCached RT

 l201/l201(Unspecified)D   N
 0UNKNOWNCached RT

 m201/s  92.x.x.95  D   N
 54020OK (32 ms) Cached RT

 n201   (Unspecified) 

Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/24 Dan Journo d...@keshercommunications.com

  I am curious, is your version of asterisk correctly compiling the
 regserver field? Each server needs to have a distinct server name.

 ** **

 Upgrading to the latest version didn't help. After about 30 minutes,
 Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
 Registered on Asterisk1.


It is something really amazing... Can you run sip show peers on each one
of the servers and post the response?

You said the second asterisk is completely opaque to your peers. Can you
run a tcpdump on secondary server to see if for some obscure reason the
phones try to contact the secondary asterisk?

Leandro
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/23 Dan Journo d...@keshercommunications.com

 Hi,

 ** **

 We're trying to decide whether to switch back to a static file for
 sip.conf. Currently we use mysql realtime but can't see any real benefit.*
 ***

 ** **

 Why would someone choose realtime sip over static files?

 ** **

 Thanks

 ** **

 Dan Journo

 Kesher Communications (UK)

 Business Phone Systems http://www.keshercommunications.com/ | Hosted 
 PBXhttp://www.keshercommunications.com/hostedpbx.html
 

 T: 0161 820 8353



All depends by the number of sip peers and the number of addition/deletion
you make. If you have static files, you have to sip reload every time you
add/remove a peer. With realtime is all realtime. I have switched to
realtime peers some times ago with great benefit.

Leandro
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/23 Dan Journo d...@keshercommunications.com

  We have never experienced that and use realtime with multiple asterisk
 servers.

 We've only recently started seeing the problem.

 To simplify the issue, assuming we have two servers, Asterisk1 and
 Asterisk2...

 Asterisk1 is a primary server and Asterisk2 is a backup and used as a
 failover. Asterisk is running on Asterisk2 to speed up the switch.
 Both share the realtime database.

 For some reason, about 5% of sip peers are listed as Registered on
 Asterisk2 even though there is no way they could discover the IP of
 Asterisk2 on their own.
 They also happen to be registered on Asterisk1 where they are supposed to
 be.

 SIP Debug has shown that they aren't actually registering with Asterisk2
 at all. They are only sending OPTIONS keepalive messages to Asterisk2 since
 QUALITY=yes something.
 They never actually send a REGISTER to Asterisk2 so that server must be
 picking up the Peer status from the realtime DB.


I have multiple asterisk servers with a pure 100% realtime configuration.
They are all working together and sharing the same realtime database (not
only sipfriends, but queue, voicemail, meetme, musiconhold and others)
without any of the problems you have reported.

Maybe you are lacking some of the configuration. These is the relevant part.

rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/23 Dan Journo d...@keshercommunications.com

  Maybe you are lacking some of the configuration. These is the relevant
 part.

 ** **

  rtcachefriends=yes 

  rtsavesysname=yes

  rtupdate=yes  

  rtautoclear=yes  

 ** **

 We have 

 rtcachefriends=yes 

 rtsavesysname=yes

 ** **

 and these we don't have but they are set to YES by default

  rtupdate=yes  

  rtautoclear=yes  

 ** **

 Its probably an issue with the version of Asterisk we are using because I
 haven't had this problem in the past.


I am running the latest 1.8 version. Which version are you running?

Leandro
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/24 Dan Journo d...@keshercommunications.com

  Its probably an issue with the version of Asterisk we are using
 because I haven't had this problem in the past.

  I am running the latest 1.8 version. Which version are you running?

 ** **

 ** **

 1.8.15.0. I'll upgrade it to 1.8.20.1 when I can and see if it makes a
 difference.

 --


I am curious, is your version of asterisk correctly compiling the regserver
field? Each server needs to have a distinct server name.

Leandro
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Re: [asterisk-users] [SOLVED] Blind transfer behavior - Asterisk 1.8 and 10

[Leandro Dardini]

Can you please post a dialplan excerpt about using these variables. I just
tried using them, but they are all empty. Maybe I am making the same
mistake of you.

Leandro

2013/1/22 Administrator TOOTAI ad...@tootai.net

 Please forget this message, BLINDTRANSFER is working, I had a typo in the
 dialplan when using this variable.

 Apologize

 Le 22/01/2013 10:40, Administrator TOOTAI a écrit :

  Hi,

 I want to check the status of a blind transfer (only sip endpoint)
 between various phones. Transfer is working perfectly, using ## from
 features.conf or using transfer key from phone, here SNOM320.

 My problem is that if party to transfer to is busy, the transfer fail
 and the call is ended. What I want to do is to return the call to the
 party who originate the transfer.

 I checked variable like ${BLINDTRANSFER} ${TRANSFERED_BY} or
 ${TRANSFER_CONTEXT}, they are all empty. What did I miss?

 Thanks for any hints


 --
 Daniel

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Re: [asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk

[Leandro Dardini]

2013/1/17 Onur Cem Çelebi occel...@gmail.com

 Hello,

 My problem is, outgoing calls (from asterisk to CCM) work fine but
 incoming (from CCM to Asterisk) does not work because of CCM is trying to
 use g729 over SIP trunk. I have found that link after a quick search.
 Problem is the same as in link below (However my Asterisk version is
 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for
 using g729 codec. The post was written in 2006. Is there any better
 solution since that time ? Thanks for reading.

 link : g279 codec over SIP Trunk between CCM and 
 Asteriskhttps://supportforums.cisco.com/message/1072037



Have you checked if the problem is the license? Asterisk doesn't have a
free encoder/decoder for g729, only pass through is available. Try to debug
the SIP call to see if the capabilities don't match or just buy a $10
license from Digium (1 concurrent call).

Leandro
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