[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 :

> 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  -
>  Datum: Thu, 15 Sep 2016 18:06:14 +0200
>Von: Leandro Dardini 
> 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  -
>
>
>
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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"  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 :

> 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&SIP-103) ...

[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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>>>
>>
>>
>
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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 :

> 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
# =>
,[/],[,[,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 :

> 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" 
> > 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/104&SIP/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 :

> 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  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 :

>
> 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 
> :
>
>
>> 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 :

> 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 :

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

[Leandro Dardini]

I love you all

:-)

Leandro


2014-02-05 Richard Mudgett :

>
>
>
> On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini wrote:
>
>> 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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[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] 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 :

>  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

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

[Leandro Dardini]

2014/1/23 Matthew Jordan 

> On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini 
> 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
>
> --
> Matthew Jordan
> 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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[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] 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 

>  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 
>
>>  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/123&SIP/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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>
>
>
>
>
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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 

>  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/123&SIP/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] 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 

> 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  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
>
>
>
> --
> _
> 

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 

> 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/123&SIP/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/123&SIP/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] 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 

> Is directmedia set to no?
>
>
> On 15 January 2014 23:11, Leandro Dardini  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 

> Thanks AJ,
> The capturing of agent activities on their desktop by the supervisor.
> Regards
> On 20 Dec 2013 12:18, "A J Stiles"  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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>
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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 

> 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 

> 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" ;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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[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] 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 

> 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 wrote:
>
>> 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 
>>
>>> 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
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocat

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 

> 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
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk Realtime Static Voicemail

[Leandro Dardini]

2013/11/11 John T. Bittner 

>  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.*
>
>
>
> --
>
>
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] Passcode

[Leandro Dardini]

Again, the authenticate function can help you

Leandro


2013/5/20 Felix Vazquez 

>  How do I make a user dial a passcode if he wants to make an
> international call?
>
>
>
> --
>
> This electronic message contains information from BOSH Global Services
> which may be company sensitive, proprietary, privileged or otherwise
> protected from disclosure. The information is intended to be used solely by
> the recipient(s) named above. If you are not an intended recipient, be
> aware that any review, disclosure, copying, distribution or use of this
> transmission or its contents is prohibited. If you have received this
> transmission in error, please notify the sender immediately.
>
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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 

>  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.
>
>
>
> --
>
> This electronic message contains information from BOSH Global Services
> which may be company sensitive, proprietary, privileged or otherwise
> protected from disclosure. The information is intended to be used solely by
> the recipient(s) named above. If you are not an intended recipient, be
> aware that any review, disclosure, copying, distribution or use of this
> transmission or its contents is prohibited. If you have received this
> transmission in error, please notify the sender immediately.
>
> --
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Re: [asterisk-users] Loopback question

[Leandro Dardini]

Is the "echo" application suitable to you?

Leandro


2013/5/20 CDR 

> 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] 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 

> 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 
> *To:* Tommy Cooper ; Asterisk Users Mailing List -
> Non-Commercial Discussion 
> *Sent:* Thursday, April 18, 2013 11:04 PM
> *Subject:* Re: [asterisk-users] Dynamic realtime + queues
>
>  You need a "name" column. This is

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 

> 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
> *

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"  ha
scritto:

> - Original Message -
> > From: "Leandro Dardini" 
> > 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 " 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  load"
> to load it back from the db.
>
> Michael
>
> --
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[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] 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"  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 :
> 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]

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 :
> 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 
> 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 
>> 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 
>>> 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 
>>>> 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 

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 :
> 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 
> 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  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
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
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>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] trunking trixbox - panasonic

[Leandro Dardini]

2013/3/12 kepin sinatra 

> hi, anybody know how to make a sip trunk between trixbox and panasonic kx
> tde 100?
> i've tried, but always failed when calling from trixbox to panasonic.
>
> thanks,,,
>
>
> --
>

Try a SIP debug to understand the reason it fails. Is it a problem of
codec? Is it a problem of license?

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

[Leandro Dardini]

2013/3/8 nik600 

> 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] 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] asterisk with 1000 extensions

[Leandro Dardini]

2013/3/7 Duncan Turnbull 

>
> On 7/03/2013, at 9:29 PM, Kamlesh Kumar  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=292&CateId=147&BaseInfoCateId=292&Cate_Id=292&parentcateid=147?ProductsCateID=292&CateId=147&BaseInfoCateId=292&Cate_Id=292&parentcateid=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] asterisk with 1000 extensions

[Leandro Dardini]

2013/3/7 Steve Edwards 

> 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] 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 

> 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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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 

> Hi,
>
> You might want to use ${MACRO_EXTEN} variable inside to preserve exten
> variable of the original dialplan exten variable.
>
> Mitul
> On Feb 24, 2013 4:04 PM, "Leandro Dardini"  wrote:
>
>> I just discover an "hidden" problem with AEL macro I want to have your
>> feedback. If you use a macro to dial out, like &dialout(${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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[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] 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 

> 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
>
>
> --
> --
> 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] Playback on h exten

[Leandro Dardini]

2013/2/21 Enrico Pasqualotto 

> 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
> --  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] Remove Abandoned call

[Leandro Dardini]

2013/2/21 akhilesh chand 

> 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] Asterisk question

[Leandro Dardini]

2013/2/20 Nguyễn Công 

> 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" 
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 
> Date: Mon, February 04, 2013 12:25 pm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
>
>  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.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 
> Date: Mon, February 04, 2013 12:05 pm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
>
>
> 
> **
>  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*),
> I see this example:
>   $fp = fsockopen("www.example.com", 80, $errno, $errstr, 30);
> if (!$fp) {
> echo "$errstr ($errno)\n";
> } else {
> $out = "GET / HTTP/1.1\r\n";
> $out .= "Host: *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*[
> *mailto: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 

> On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
> > On Wed, Jan 30, 2013 at 12:05 PM, XBrian  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] 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 

> 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] Asterisk Messaging Refuses To Work!

[Leandro Dardini]

2013/1/30 XBrian 

> 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] 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 

> 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"  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]

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)D   N
> 0UNREACHABLE Cached RT
>
> ** **
>
> *Show peers Output from a secondary server*
>
> Name/username  HostDyn
> Forcerport ACL Port Status Realtime
>
> d20194.x.x.228 D   N
> 5060 UNREACHABLE Cached RT
>
> e202  94.x.x.44 D
> N 55022UNREACHABLE Cached RT
>
> e203  94.x.x.44 D
> N 55024UNREACHABLE Cached RT
>
> e204  94.x.x.44 D
> N 55008UNREACHABLE Cached RT
>
> e205  94.x.x.44 D
>  N 55016UNREACHABLE Cached RT
>
> e206  94.x.x.44 D
> N 55014UNREACHABLE Cached RT
>
> e207  94.x.x.44 D
> N 55020UNREACHABLE Cached RT
>
> e209  94.x.x.44 D
> N 55012UNREACHABLE Cached RT
>
> e210  94.x.x.44 D
> N 55010UNREACHABLE Cached RT
>
> e212  81.x.x.93D
> N 5060 UNREACHABLE Cached RT
>
> h201   217.x.x.78   D   N
> 38980UNREACHABLE Cached RT
>
> j201 94.x.x.62D   N
> 57813UNREACHABLE Cached RT
>
> o201 92.x.x.86 D
> N 51824OK (47 ms) Cached RT
>
> o202 92.x.x.86 D
> N 58641OK (47 ms) Cached RT
>
> o203 92.x.x.86 D
> N 49172OK (48 ms) Cached RT
>
> o204 (Unspecified)D
> N 0UNREACHABLE Cached RT
>
> k201   2.x.x.169 D   N
> 52757OK (52 ms) Cached RT
>
> k202   195.x.x.5 D   N
> 20569UNREACHABLE Cached RT
>
> ** **
>
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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...

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

[Leandro Dardini]

2013/1/24 Dan Journo 

> > 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]

2013/1/24 Dan Journo 

> >> 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] Realtime vs Static Files

[Leandro Dardini]

2013/1/23 Dan Journo 

> > 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]

2013/1/23 Dan Journo 

> > 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 

> 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 
> PBX<http://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] [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 

> 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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