from:"Social Boh"

Re: [asterisk-users] ICTBroadcast Version 4.2 released

2021-08-14 Thread Social Boh


BLA BLA BLA... liar

---
I'm SoCIaL, MayBe

El 14/08/2021 a las 6:38 a. m., Tahir Almas escribió:
Sorry ,  posted here being it as asterisk based project,  it would not 
happen again


regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com <http://www.ictinnovations.com>
Leveraging open source in ICT


On Sat, Aug 14, 2021 at 4:29 PM Social Boh <mailto:soc...@bohboh.info>> wrote:


He do this always y still have access to this list.

Very bad behaviour.

---
I'm SoCIaL, MayBe

El 14/08/2021 a las 4:50 a. m., Antony Stone escribió:
> On Saturday 14 August 2021 at 11:43:55, Tahir Almas wrote:
>
>> Pleased to announce the release of asterisk based auto dialer
and call
>> center software solution
> Please note that you have posted this advertisement to the
mailing list
> "Asterisk Users - Non-Commerical Discussion".
>
>
> Regards,
>
>
> Antony.
>

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Re: [asterisk-users] ICTBroadcast Version 4.2 released

2021-08-14 Thread Social Boh


He do this always y still have access to this list.

Very bad behaviour.

---
I'm SoCIaL, MayBe

El 14/08/2021 a las 4:50 a. m., Antony Stone escribió:

On Saturday 14 August 2021 at 11:43:55, Tahir Almas wrote:


Pleased to announce the release of asterisk based auto dialer and call
center software solution

Please note that you have posted this advertisement to the mailing list
"Asterisk Users - Non-Commerical Discussion".


Regards,


Antony.



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Re: [asterisk-users] [SOLVED] CentOS 7 yum-update and Gotoif has stoppped to workink

2021-02-05 Thread Social Boh


Thank you

solved with

*yum history undo ID*

and then:

*yum update --exclude=glibc**

https://www.voztovoice.org/?q=node/2901

---
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El 5/02/2021 a las 11:26 a. m., Michael L. Young escribió:


- On Feb 5, 2021, at 11:18 AM, Michael L. Young 
 wrote:


- On Feb 4, 2021, at 4:26 PM, Social Boh 
wrote:

The problem is with this CentOS 7 glibc version:

2.17-317.el7

After the library update and system reboog, gotoif Asterisk
application, stop to working

Any hint to solve?

Until it is resolved, you can do a 'yum history' and note the
transaction ID of the update.  Then try running 'yum history undo
[transaction id]'.  That should roll you back to the previous glibc.

Looks like Red Hat is already working on it:
  https://access.redhat.com/solutions/5778071

Here is the Bugzilla report for anyone on RHEL / CentOS 
7: https://bugzilla.redhat.com/show_bug.cgi?id=1925204


--
Michael Young
(elguero)

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Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

2021-02-04 Thread Social Boh


The problem is with this CentOS 7 glibc version:

2.17-317.el7

After the library update and system reboog, gotoif Asterisk application, 
stop to working


Any hint to solve?

---
I'm SoCIaL, MayBe

El 4/02/2021 a las 11:43 a. m., Social Boh escribió:

Hello,

today a very strange thing happens on all my CentOS 7 servers. After a 
yum update, on all my Asterisk 16.16.0 version, the GotoIF application 
stop to working.


A gotoif example stop to working is:

same => n,GotoIf($["${ARG1:0:1}" = "+"]?no)

the result on the Asterisk console is:

 GotoIf("SIP/open1-008f", "nan?cli") in new stack

On all Gotoif on my dialplan, the result is the same... nan?cli

The yum update include this files:

gd-last x86_64 2.3.1-1.el7.remi remi-safe 
135 k

 glibc x86_64 2.17-322.el7_9 updates 3.6 M
 glibc-common x86_64 2.17-322.el7_9 
updates    12 M
 glibc-devel x86_64 2.17-322.el7_9 
updates   1.1 M
 glibc-headers x86_64 2.17-322.el7_9 
updates   690 k
 iwl105-firmware noarch 18.168.6.1-80.el7_9 
updates   234 k
 iwl135-firmware noarch 18.168.6.1-80.el7_9 
updates   243 k
 iwl2000-firmware noarch 18.168.6.1-80.el7_9 
updates   237 k
 iwl2030-firmware noarch 18.168.6.1-80.el7_9 
updates   245 k
 iwl3160-firmware noarch 25.30.13.0-80.el7_9 
updates   1.5 M
 iwl6000g2b-firmware noarch 18.168.6.1-80.el7_9 
updates   305 k
 iwl7260-firmware noarch 25.30.13.0-80.el7_9 
updates   6.1 M
 kernel-headers x86_64 3.10.0-1160.15.2.el7 
updates   9.0 M
 kernel-tools x86_64 3.10.0-1160.15.2.el7 
updates   8.1 M
 kernel-tools-libs x86_64 3.10.0-1160.15.2.el7 
updates   8.0 M
 libblkid x86_64 2.23.2-65.el7_9.1 
updates   183 k
 libmount x86_64 2.23.2-65.el7_9.1 
updates   185 k
 libsmartcols x86_64 2.23.2-65.el7_9.1 
updates   143 k
 libuuid x86_64 2.23.2-65.el7_9.1 
updates    84 k
 libuuid-devel x86_64 2.23.2-65.el7_9.1 
updates    93 k
 linux-firmware noarch 20200421-80.git78c0348.el7_9 
updates    80 M
 perl x86_64 4:5.16.3-299.el7_9 updates   
8.0 M
 perl-ExtUtils-Embed noarch 1.30-299.el7_9 
updates    51 k
 perl-ExtUtils-Install noarch 1.58-299.el7_9 
updates    75 k
 perl-Pod-Escapes noarch 1:1.04-299.el7_9 
updates    52 k
 perl-devel x86_64 4:5.16.3-299.el7_9 
updates   454 k
 perl-libs x86_64 4:5.16.3-299.el7_9 
updates   690 k
 perl-macros x86_64 4:5.16.3-299.el7_9 
updates    44 k
 php70-php-common x86_64 7.0.33-25.el7.remi 
remi-safe 597 k
 php70-php-json x86_64 7.0.33-25.el7.remi 
remi-safe  66 k
 php70-php-xml x86_64 7.0.33-25.el7.remi 
remi-safe 173 k
 php70-php-xmlrpc x86_64 7.0.33-25.el7.remi 
remi-safe  83 k
 php71-php-common x86_64 7.1.33-12.el7.remi 
remi-safe 609 k
 php71-php-json x86_64 7.1.33-12.el7.remi 
remi-safe  67 k
 python-perf x86_64 3.10.0-1160.15.2.el7 
updates   8.1 M
 systemd x86_64 219-78.el7_9.3 updates   
5.1 M
 systemd-libs x86_64 219-78.el7_9.3 
updates   418 k
 systemd-sysv x86_64 219-78.el7_9.3 
updates    97 k

 tuned noarch 2.11.0-11.el7_9 updates   269 k
 util-linux x86_64 2.23.2-65.el7_9.1 
updates   2.0 M

 zlib x86_64 1.2.7-19.el7_9 updates 90 k
 zlib-devel x86_64 1.2.7-19.el7_9 
updates    50 k



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[asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

2021-02-04 Thread Social Boh


Hello,

today a very strange thing happens on all my CentOS 7 servers. After a 
yum update, on all my Asterisk 16.16.0 version, the GotoIF application 
stop to working.


A gotoif example stop to working is:

same => n,GotoIf($["${ARG1:0:1}" = "+"]?no)

the result on the Asterisk console is:

 GotoIf("SIP/open1-008f", "nan?cli") in new stack

On all Gotoif on my dialplan, the result is the same... nan?cli

The yum update include this files:

gd-last x86_64 2.3.1-1.el7.remi remi-safe 135 k
 glibc x86_64 2.17-322.el7_9 updates   3.6 M
 glibc-common x86_64 2.17-322.el7_9 
updates    12 M
 glibc-devel x86_64 2.17-322.el7_9 
updates   1.1 M
 glibc-headers x86_64 2.17-322.el7_9 
updates   690 k
 iwl105-firmware noarch 18.168.6.1-80.el7_9 
updates   234 k
 iwl135-firmware noarch 18.168.6.1-80.el7_9 
updates   243 k
 iwl2000-firmware noarch 18.168.6.1-80.el7_9 
updates   237 k
 iwl2030-firmware noarch 18.168.6.1-80.el7_9 
updates   245 k
 iwl3160-firmware noarch 25.30.13.0-80.el7_9 
updates   1.5 M
 iwl6000g2b-firmware noarch 18.168.6.1-80.el7_9 
updates   305 k
 iwl7260-firmware noarch 25.30.13.0-80.el7_9 
updates   6.1 M
 kernel-headers x86_64 3.10.0-1160.15.2.el7 
updates   9.0 M
 kernel-tools x86_64 3.10.0-1160.15.2.el7 
updates   8.1 M
 kernel-tools-libs x86_64 3.10.0-1160.15.2.el7 
updates   8.0 M
 libblkid x86_64 2.23.2-65.el7_9.1 
updates   183 k
 libmount x86_64 2.23.2-65.el7_9.1 
updates   185 k
 libsmartcols x86_64 2.23.2-65.el7_9.1 
updates   143 k
 libuuid x86_64 2.23.2-65.el7_9.1 
updates    84 k
 libuuid-devel x86_64 2.23.2-65.el7_9.1 
updates    93 k
 linux-firmware noarch 20200421-80.git78c0348.el7_9 
updates    80 M

 perl x86_64 4:5.16.3-299.el7_9 updates   8.0 M
 perl-ExtUtils-Embed noarch 1.30-299.el7_9 
updates    51 k
 perl-ExtUtils-Install noarch 1.58-299.el7_9 
updates    75 k
 perl-Pod-Escapes noarch 1:1.04-299.el7_9 
updates    52 k
 perl-devel x86_64 4:5.16.3-299.el7_9 
updates   454 k
 perl-libs x86_64 4:5.16.3-299.el7_9 
updates   690 k
 perl-macros x86_64 4:5.16.3-299.el7_9 
updates    44 k
 php70-php-common x86_64 7.0.33-25.el7.remi 
remi-safe 597 k
 php70-php-json x86_64 7.0.33-25.el7.remi 
remi-safe  66 k
 php70-php-xml x86_64 7.0.33-25.el7.remi 
remi-safe 173 k
 php70-php-xmlrpc x86_64 7.0.33-25.el7.remi 
remi-safe  83 k
 php71-php-common x86_64 7.1.33-12.el7.remi 
remi-safe 609 k
 php71-php-json x86_64 7.1.33-12.el7.remi 
remi-safe  67 k
 python-perf x86_64 3.10.0-1160.15.2.el7 
updates   8.1 M

 systemd x86_64 219-78.el7_9.3 updates   5.1 M
 systemd-libs x86_64 219-78.el7_9.3 
updates   418 k
 systemd-sysv x86_64 219-78.el7_9.3 
updates    97 k

 tuned noarch 2.11.0-11.el7_9 updates   269 k
 util-linux x86_64 2.23.2-65.el7_9.1 
updates   2.0 M

 zlib x86_64 1.2.7-19.el7_9 updates    90 k
 zlib-devel x86_64 1.2.7-19.el7_9 
updates    50 k



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Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

2020-03-23 Thread Social Boh


Because Asterisk do not support PUBLISH.

For BLF Configuration:

https://wiki.asterisk.org/wiki/display/AST/Configuring+chan_sip+for+Presence+Subscriptions

or

https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+for+Presence+Subscriptions

---
I'm SoCIaL, MayBe

On 3/23/20 05:13, John Hughes wrote:
Hi, in these dark days of COVID-19 lockdown I'm using linphone to 
connect to my office asterisk system for working from home.


It's going pretty well but the presence/BLF functions don't appear to 
work.


In the linphone logs and asterisk debug I find that asterisk is 
rejecting linphone's PUBLISH message:


<--- SIP read from UDP:10.27.128.3:5060 --->
PUBLISH sip:j...@xxx.xxx.com SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport
From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com
CSeq: 20 PUBLISH
Call-ID: SMHLUSLJD6
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Accept: application/pidf+xml
Content-Length: 511
Content-Type: application/pidf+xml
Expires: 3600
User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)


xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" 
xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd; 
entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> 
  open  
 sip:j...@xxx.xxx.com 
2020-03-23T09:40:43Z 


<->
--- (14 headers 3 lines) ---


Sending to 10.27.128.3:5060 (no NAT)

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060

From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com;tag=as674d428f
Call-ID: SMHLUSLJD6
CSeq: 20 PUBLISH
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0

I can find nothing in the asterisk logs that says *why* it doesn't 
like the publish.


Help?




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Re: [asterisk-users] configure SRTP port range?

2019-02-23 Thread Social Boh

*DIrect media with SRTP is not supported. All media when SRTP goes 
through Asterisk.*


So you have to open ports on your firewall and disable directmedia=yes 
on your configuration.


Only open a range of ports that you really use: for example is you have 
maximum 10 simultaneous calls, open only 40 ports (4 ports for each 
call, two for RTP and two for RTCP). Then change rtp.conf configuration 
reflect the range of ports you using.


Other option is using another PBX/SWITCH that support SRTP flow direct 
between endpoints.


Regards

---
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Re: [asterisk-users] Hint and state

2019-01-10 Thread Social Boh


Hello,

maybe:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_DEVICE_STATE

---
I'm SoCIaL, MayBe

El 10/01/2019 a las 09:13, Administrator TOOTAI escribió:

Hi,

on an Asterisk 16 with PJSIP I want to know the state of a device 
(idle, busy, unavailable, ...) in the dialplan. I tried with 
ChanIsAvail() but this one doesn't return the real state (eg a device 
calling an extension which is running ChanIsAvail() is marked as idle!)


When I use in a console "core show hints" or "core show hint 
" I get the right information. How to get the same 
information in a dialplan ?


Thanks for any hint (ha ha ;))

Daniel



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Re: [asterisk-users] Pjsip and Call limit

2018-12-27 Thread Social Boh


Hello,

ringinuse still working on queue configuration:

ringinuse=no

Ring the extension only if not in use.

Regards

---
I'm SoCIaL, MayBe

El 27/12/2018 a las 17:11, Administrator TOOTAI escribió:

Le 27/12/2018 à 20:42, Social Boh a écrit :

Hello,

you have to use GROUP and GROUP_COUNT functions.


Well, could be done for extensions but for queue ? Does it mean 
ringinuse is useless ?



[...]
El 27/12/2018 a las 14:14, Administrator TOOTAI escribió:

Hello,

I'm used to set call-limit in sip.conf Now I switched one customer 
Asterisk to 16 version and can't get the behavior back, as well for 
extensions as for queues.


I set ringinuse=no for queues and have max_audio_streams = 1 
max_video_streams = 0. I wanted to add max_calls = 1 but this 
parameter is not accepted.


Thanks for any hint



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Re: [asterisk-users] New features released in ICTBroadcast

2018-12-19 Thread Social Boh

Please STOP send this kind of messages. Use only 
asterisk-...@lists.digium.com list


Thank you

---
I'm SoCIaL, MayBe

El 19/12/2018 a las 06:00, Tahir Almas escribió:
Following new features  are now  supported by asterisk based 
telemarketing  software


Auto subscription / registration after call recipient press a key in 
voice broadcasting


https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer

There will be restriction to call a  number  in off time  accordingly  
to timezone of  destination number automatically


https://www.ictbroadcast.com/Time-Zone-based-restrictions-on-telemarketing-campaigns-ICTBroadcast-autodialer-scheduling

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




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Re: [asterisk-users] Capture SIP all the time

2018-12-05 Thread Social Boh


sipdebug = yes

sip.conf

---
I'm SoCIaL, MayBe

El 05/12/2018 a las 17:11, Saint Michael escribió:
Is there a way to configure the old SIP channel to stay in sip set 
debug all the time, without human intervention and also at boot time, 
by default?






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Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?

2018-07-28 Thread Social Boh


TIMEOUT function:

example

same => n,Set(TIMEOUT(absolute)=600)

after 600 seconds Asterisk Hankup the call

Regards

---
I'm SoCIaL, MayBe

On 7/28/18 16:08, Jonathan H wrote:

Last question for today, I promise!

The problem: In order to disconnect calls after x minutes, I need to do this:

[setup]
exten => setup,1,Answer()
 same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
 same => 
n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
 same => n,Dial(Local/s@root/n,3,L(354:6))
 same => n,Hangup()

[root]
exten => s,1,Verbose(1,Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)

etc etc

Works well, but the result is it looks like there are 2 active calls
in the console. Is there any way of forcing the drop of a call after x
minutes without doing this "double dialling" business?

Thanks




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Re: [asterisk-users] PJSIP issue - Syntax error exception when parsing

2018-02-21 Thread Social Boh

hello,

you receive this error because the second line of SIP message not can
begin without a Header. You Phone or server (maybe OpenSIPs or Kamailio)
whet quitting a Via Header make some kind of error so the result is you
have the Via Header in two lines instead one.

Regards

---
I'm SoCIaL, MayBe

El 21/02/2018 a las 03:39, Michele Pinassi escribió:
> Hi all, i'm getting this error:
>
> [Feb 21 09:29:09] ERROR[1250]: pjproject:0 :       
> sip_transport.c Error processing 396 bytes packet from UDP
> 193.x:5060 : PJSIP syntax error exception when parsing '' header on
> line 2 col 1:
> SIP/2.0 480 User 7000 not registered
>
> Via: SIP/2.0/UDP
> 193.x:5060;received=193.xx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29
> From: ;tag=3d0a19e7-eabe-4446-84dd-43f02d831033
> To: ;tag=24eb447e8d0b8b1e81ba6efb9d8649a2.ea35
> Call-ID: defee3c7-e5ba-41ff-9be7-3c37e62437f2
> CSeq: 22011 INVITE
> Content-Length: 0
>
>
> -- end of packet.
>
> Asterisk 15.2.0 and PJSip 2.7.1
>
> Tnx, Michele
>
>
>

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