Re: [asterisk-users] Mailing List Future
On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > To that end, we’ve decided to discontinue the mailing lists effective > February 1st, 2024. That's a very sad news! :-( -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio from soft phone actual phone from cloud
On Wed, 2023-07-19 at 12:42 -0400, Jerry Geis wrote: > Why might I not be getting audio ? Make sure the RTP port range is correctly configured and open on your server's firewall. The port range is defined in /etc/asterisk/rtp.conf The same range of UDP ports must be correctly forwarded on your firewall from the outside to Asterisk. For example, in rtp.conf: [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=10002 rtpend=10199 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intro and question
Even I was confused, and the directions in that book seem like a complication of a simple affair, at least for my modest needs. Finally, I installed Asterisk with apt and created extensions.conf and pjsip.conf files. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() after the h extension has been invoked?
On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > I use Dial() commands with custom SIP headers to pass information > (eg: about > the current state of a call) between the front-end and back-end > machines, and > this works very well. > > I need to perform a Dial() > command after an inbound channel has hung up. I do not expect the > Dial() to > bridge to anything (the context being dialled simply does some > database > manipulation and then hangs up without even bothering to answer). > > > Any suggestions welcome :) Maybe you can use the "g" option in the first Dial(...) and proceed in the dial plan with the second Dial(...) g - Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up. Example: exten => 1234,1,Dial(SIP/deskphone,120,g) same => n,Dial(SIP/cordlessphone) same => n,Hangup() Extension 1234 dials a deskphone. If "deskphone" answer... bla bla bla... and after "deskphone" hangs up, the "cordlessphone" is dialed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notifying missed calls
On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote: > Really, I can't understand what you mean... I'm feeling really > dumb... No need to feel dumb. I'm not an expert and when I look to my extensions.conf... well... countless pulling my hairs out, head banging on the keyboard,,, :-) The "h" extension is executed whenever a call is hang up in that contexts. In your configuration it executes first the "s" extension (where you GoTo h,1) and once that is executed, the "h" extension is executed again. Take a look to the example I posted. It's very basic, but it does the job. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notifying missed calls
On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > 1) The E-Mails will be sent "double" It sends the first mail by executing "noanswer,2" and a second mail because because of "main-incoming,h,2" > 2) The E-Mails will be sent for outgoing unanswered calls, too. Use the "h" extension only in the context for incoming calls > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notifying missed calls
Here my configuration:
[incoming]
; Incoming from Swisscom
exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)})
same => n,Dial(SIP/deskphone,120)
same => n,Hangup()
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s
"Missed Call from ${CALLERID(num)}" my-em...@address.here)
exten => h,n(done),NoOp()
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Re: [asterisk-users] Detect if people is talking
On Wed, 2020-12-30 at 12:09 -0300, Valter Nogueira wrote: > Is there any way to detect if an agent is speaking? https://wiki.asterisk.org/wiki/display/AST/Application_WaitForSilence https://wiki.asterisk.org/wiki/display/AST/Application_WaitForNoise -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which linux for asterisk?
On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote: > what is best choice ? Oracle? Ubuntu? I'm running Asterisk since several years on Ubuntu without any issues. Debian should be fine too. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?
On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP account details > and now require usage of their own ”router’s”. That's very irritating and make me angry. Few of my client had the same problem. The solution: write a letter asking the SIP credentials explaining you want configure your own equipment and tell them you switch to another provider in case of refusal. Good luck! I don't know if there is an appropriate hardware to build a DECT bridge and I doubt that fiddling with anything like that will not be a reliable solution. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail2Fax
On Wed, 2020-06-17 at 18:10 +0200, basti wrote: > txfax seem to be a port of spandsp. it is also old. > Is there a newer way to send fax via asterisk. I don't know if it's newer, but I use "sendfax" -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] email notification on missed call
On Sat, 2019-11-02 at 11:42 +0100, Antony Stone wrote:
> Doesn't that send an email for every call once it ends, not just
> unanswered ones?
Whoops! You are right! :-)
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call Open on Asterisk from
${CALLERID(num)}" | mail -s "Missed VIP Call on Asterisk from
${CALLERID(num)}" -a "From: Astersik PBX " myemailaddr
e...@example.com)
exten => h,n(done),NoOp()
exten => h,n,HangUp()
> > exten => h,n,HangUp()
> That looks most strange to me - calling Hangup() in the hangup
> extension...
:-D
Probably it is not necessary. But isn't a good practice to end any
extension with a "HangUp"?
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Re: [asterisk-users] email notification on missed call
On Wed, 2019-10-30 at 05:10 +0100, Fourhundred Thecat wrote:
> what is the best way to implement email notification on missed call ?
> Is there perhaps a better way to this than described above ?
This is my way:
exten => h,1,System(echo "Missed Call Open on Asterisk from
${CALLERID(num)}" | mail -s "Missed VIP Call on Asterisk from
${CALLERID(num)}" -a "From: Astersik PBX " myemailaddr
e...@example.com)
exten => h,n,HangUp()
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Re: [asterisk-users] Asterisk 13.28.0 Now Available
Thank you, dear Asterisk Development Team, for this great software! > The Asterisk Development Team would like to announce the release of > Asterisk 13.28.0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS and SIM card
On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? To send SMS, there is a CLI command. You can use the commands in your extensions.conf accordingly your needs. http://wiki.e1550.mobi/doku.php?id=usage dongle sms Send SMS to with the using -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS and SIM card
On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote:
> Any small example how to send gsm calls through chan_dognle and how
> to send sms through chan_dongle?
In dongle.conf:
[gsmgateway]
context=gsm
imei=123456789012345
imsi=098765432112345
In extensions.conf:
[gsm]
; Incoming calls from GSM/3G
exten => +41776665544,1,Dial(Local/mydeskphone@voipphone)
same => n,Hangup()
; Phone call though GSM/3G
exten => _0.,n,Dial(Dongle/gsmgateway/${EXTEN},120)
same => n,Hangup()
; Incoming SMS to mail address and to sms.txt file
exten => sms,1,Noop(Incoming SMS from ${CALLERID(num)}
${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} -
${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >>
/var/log/asterisk/sms.txt)
exten => sms,n,System(echo "${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} -
${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}" |
mail -s "SMS from ${CALLERID(num)}" myemailaddr...@mymailbox.com)
exten => sms,n,Hangup()
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Re: [asterisk-users] Sending SMS and SIM card
You can use a cheap 3G-USB-dongle and chan_dongle. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asking
On Tue, 2018-09-18 at 20:28 +0200, modou lo wrote: > Hello, please can i have a code which help me to tax user every voip > services in asterisk means when user starts to call someone Check Asterisk2billing http://www.asterisk2billing.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decoding SIP register hack
On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:
> 3. How do I set up the server to block these ?
>
> 4. Can I stop the retransmitting of the 401 Unauthorized packets ?
I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:
in /etc/asterisk/logger.conf:
messages => security,notice,warning,error
in /etc/asterisk/sip.conf:
allowguest=yes
context=unauthenticated
in /etc/asterisk/extensions.conf:
[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()
exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()
in /etc/fail2ban/jail.conf:
[asterisk]
filter = asterisk
action = iptables-allports[name=ASTERISK]
logpath = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime = 518400
enabled = true
in /etc/fail2ban/filter.d
# Fail2Ban configuration file
#
#
# $Revision: 250 $
#
[INCLUDES]
# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the
logfile. The
# host must be matched by a group named "host". The tag
"" can
# be used for standard IP/hostname matching and is only an
alias for
# (?:::f{4,6}:)?(?P\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
NOTICE.* .*: No registration for peer '.*' \(from
\)
NOTICE.* .*: Host failed MD5 authentication for
'.*' \(.*\)
NOTICE.* .*: Host denied access to register peer
'.*'
NOTICE.* .*: Host did not provide proper
plaintext password for '.*'
NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
NOTICE.* .*: Peer '.*' is not dynamic (from )
NOTICE.* .*: Host denied access to register peer
'.*'
SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
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Re: [asterisk-users] invite to conference by a call file
Maybe something like a local web page where your secretary can enter the list of phone numbers to call and a script that generates a call file and moves it to the Asterisk spool folder. But that's not an Asterisk issue. It's more a programmer's issue. :-) On Thu, 2018-03-22 at 16:06 +0200, Atux Atux wrote: > that's the problem. it is never the same people > I need the office secretary to edit a file (call file) > and place it in a particular folder in their windows PCs. this folder > is the outgoing folder of LINUX shared through samba in LAN. i need > to make it as easy as possible, please. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invite to conference by a call file
If the participants are always the same people, there is no need to change the dialplan. Just tells the office secretary "Please, place a conference call.". with the "Page" application, she picks up the phone, dials a predefined number and all the participants are called at once. Easy peasy. :-) On Thu, 2018-03-22 at 14:21 +0200, Atux Atux wrote: > All the aforementioned techniques need change everytime on the > dialplan. I need the office secretary to edit a file (call file) and > place it in a particular folder in their windows PCs. this folder is > the outgoing folder of LINUX shared through samba in LAN. i need to > make it as easy as possible, please. > > On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist@linuxista. > com> wrote: > > Here I'm using the "Page" application to make a conference call "on > > the fly". > > > > > > > > [office] > > > > exten => ,1,Dial(SIP/desk2,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk3,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk4,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk5,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk6,150) > > same => n,Hangup() > > > > ; Conference call > > exten => ,1,Answer > > exten => ,n,Page(Local/@office/@office/ > > @office/@office/@office,d) > > same => n,Hangup() > > > > -- > > ___ > > __ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > > > Check out the new Asterisk community forum at: https://community.as > > terisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.aste > risk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invite to conference by a call file
Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => ,1,Dial(SIP/desk2,150) same => n,Hangup() exten => ,1,Dial(SIP/desk3,150) same => n,Hangup() exten => ,1,Dial(SIP/desk4,150) same => n,Hangup() exten => ,1,Dial(SIP/desk5,150) same => n,Hangup() exten => ,1,Dial(SIP/desk6,150) same => n,Hangup() ; Conference call exten => ,1,Answer exten => ,n,Page(Local/@office/@office/@off ice/@office/@office,d) same => n,Hangup()-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blacklist failed attempts
On Thu, 2018-03-01 at 15:02 +0200, Atux Atux wrote:
> I have tried to implement it through fail2ban, but it doe snot seem
> to work for my asterisk implementation.
I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:
in /etc/asterisk/logger.conf:
messages => security,notice,warning,error
in /etc/asterisk/sip.conf:
allowguest=yes
context=unauthenticated
in /etc/asterisk/extensions.conf:
[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()
exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()
in /etc/fail2ban/jail.conf:
[asterisk]
filter = asterisk
action = iptables-allports[name=ASTERISK]
logpath = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime = 518400
enabled = true
in /etc/fail2ban/filter.d
# Fail2Ban configuration file
#
#
# $Revision: 250 $
#
[INCLUDES]
# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the
logfile. The
# host must be matched by a group named "host". The tag
"" can
# be used for standard IP/hostname matching and is only an
alias for
# (?:::f{4,6}:)?(?P\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
NOTICE.* .*: No registration for peer '.*' \(from
\)
NOTICE.* .*: Host failed MD5 authentication for
'.*' \(.*\)
NOTICE.* .*: Host denied access to register peer
'.*'
NOTICE.* .*: Host did not provide proper
plaintext password for '.*'
NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
NOTICE.* .*: Peer '.*' is not dynamic (from )
NOTICE.* .*: Host denied access to register peer
'.*'
SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
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Re: [asterisk-users] email when certain numbers are called
On Mon, 2018-01-15 at 14:26 +0200, Atux Atux wrote:
> [DefaultPlan]
exten => _XX,1,System(echo "Dialed number ${EXTEN} on Asterisk
from ${CALLERID(num)}" | mail -s "Dialed number ${EXTEN} on Asterisk
from ${CALLERID(num)}" -a "From: Asterisk PBX " yo
urem...@address.com)
exten => _XX,2,Dial(SIP/VoipGate/${EXTEN},120,Tt)
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Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??
> fail2ban is most useful for blocking registration attempts. I
> handle
> non-registration call attempts by allowing guests, point them to a
> jail
> context, which runs Log(WARNING,fail2ban='${CHANNEL(peerip)}') I
> set a
> fail2ban rule to match that line logged from Asterisk.
Thanks for the suggestion. Works great! :-)
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Re: [asterisk-users] SIP trunks going to the wrong context
I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider. Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register => 22:yxc...@sip.provider.zz/22 ; Trunk3 [trunk1] type=friend host=sip.provider.zz defaultuser=11 secret=qwertz canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 1 [trunk2] type=friend host=sip.provider.zz defaultuser=22 secret=asdfgh canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 2 [trunk3] type=friend host=sip.provider.zz defaultuser=33 secret=yxcvbn canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 3 In extensions.conf: [trunkincoming] exten => 11,1,GoTo(firstline,11,1) exten => 22,1,GoTo(secondline,22,1) exten => 33,1,GoTo(thirdline,33,1) [firstline] exten => 11,1,Dial(SIP/officephone,120,m) [secondline] exten => 22,1,Dial(SIP/livingroomphone,120,m) [thirdline] exten => 33,1,Dial(SIP/bedroomphone,120,m) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for the carrier that owns a particular DID
On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote: > How do I find out which carrier owns the DID in question? Try here: https://www.twilio.com/lookup -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP, NAT and STUN/ICE
On Tue, 2017-10-10 at 11:32 +0200, Frank Vanoni wrote: > On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > > > > > local_net= 192.168.254.1/24 > > It should be: > > localnet = 192.168.254.0/255.255.255.0 Whoops... local_net=192.168.254.0/24 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP, NAT and STUN/ICE
On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > local_net= 192.168.254.1/24 It should be: localnet = 192.168.254.0/255.255.255.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial an extension to modify dialplan
On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote: > exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC) > > exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB) Whoops... sorry for the typo (in the hurry of copy & paste)! exten => 2001,1,Dial(SIP/deviceA/deviceB/deviceC) exten => 2002,1,Dial(SIP/deviceA/deviceB) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial an extension to modify dialplan
Dear Digium List
First of all, I thank all of you for all the replies and the interesting
suggestions. I thank you very much. I can only learn from people like
you. :-)
I will remember all the different solutions for a future use in other
scenarios.
On Mon, 2017-05-08 at 16:35 +0200, Frank Vanoni wrote:
> Is there a better solution?
At the end, I cleaned up my dial plan by removing the previous mess and
I'm using now ASTDB, as suggested, in the following way:
exten => 4000,1,Set(DB(alldevices/status)=OFF)
exten => 4000,2,Playback(service)
exten => 4001,1,Set(DB(alldevices/status)=ON)
exten => 4001,2,Playback(service)
exten => 2000,1,GotoIf($[${DB(alldevices/status)}=ON]?2001,1:2002,1)
exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC)
exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB)
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[asterisk-users] Dial an extension to modify dialplan
Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I did... In extensions.conf: -- snip - [mynicecontext] #include "ringdevice.conf exten => 2000,1,GoTo(ringdevice,ring,1) exten => 4000,1,System(/bin/cat /etc/asterisk/twodevices.txt > /etc/asterisk/ringdevice.conf) exten => 4000,2,Wait(3) exten => 4000,3,System(/usr/sbin/asterisk -rx "dialplan reload") exten => 4000,4,Playback(service) exten => 4001,1,System(/bin/cat /etc/asterisk/alldevices.txt > /etc/asterisk/ringdevice.conf) exten => 4001,2,Wait(3) exten => 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload") exten => 4001,4,Playback(service) -- end snip - twodevices.txt contains exten => ring,1,Dial(SIP/deviceA) alldevices.txt contains exten => ring,1,Dial(SIP/deviceA/deviceC) By dialing 4000 or 4001, the dialplan is modified and reloaded accordingly. Is there a better solution? Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log incoming calls without answering
On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote: > Any idea? I used to play with an analog telephone line and Asterisk by using a Linksys SPA-3102 Voice Gateway. I think it is no longer manufactured, but maybe you con buy a used one on eBay or you can find an equivalent device from another manufacturer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disallow CALLS without registry
On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote: > > sip.conf configuration > > In the [general] section, define: > > [general] > > ... > > allowguest=no > > alwaysauthreject=yes > > ... > With the above configuration on my Asterisk, I obtain the following result: - if the phone is registered to Asterisk, I can place any call according to the dial plan. - if the phone is NOT registered and I try to place a call, the phone obtains a "403 forbidden" at any calling attempt. Now, English is not my native language, but as far as I can understand, "forbidden" means "not allowed" or "disallowed". -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disallow CALLS without registry
On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote: > so the main question is -- how to Disallow CALLS without registering > on PBX sip.conf configuration In the [general] section, define: [general] ... allowguest=no alwaysauthreject=yes ... The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP providers connect as a guest user, however, so this may be inappropriate for your situation. Also, if you want to accept anonymous SIP calls, this line would block them, so you wouldn't want that. But it is listed here because it is the safest configuration. The "alwaysauthreject" line is important. This causes a hacker to get the same response from your PBX when they try to guess passwords whether or not they guessed a valid username. This also has the side-effect of making poorly written scanning scripts (the vast majority of hacker scripts seem to be poorly written) take less resources on your Asterisk box, as even if they scan a valid username, they'll think it doesn't exist. (Source: https://www.voip-info.org/wiki/view/Asterisk+security ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK
Hi Chris On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? Yes, it is possible! :-) A cheap solution is using a 3G-UBS-dongle. I have two SIM cards working in my Asterisk. I'm using two 3G-dongles (one for each SIM), a Huawei E173 (firmware 11.126.85.00.209) and a Huawei E180 (firmware 11.126.10.01.68). Google for "asterisk chan dongle" and you will find plenty of infos. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missed call notification
On Mon, 2016-11-28 at 14:31 +0100, tux john wrote:
> Hi. i am running asterisk 11 in debian and i would like have a missed
> call notification down to extension level.
> so if i get a missed call to extension 6589 then send an email to the
> user's email address with a subject and a text message.
> is there a guide on how to create something like that?
>From my extensions.conf
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s
"Missed Call from ${CALLERID(num)}" myemailaddr...@myemailprovider.com)
exten => h,n(done),NoOp()
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Re: [asterisk-users] Blacklist callers from file
On Sat, 2016-08-27 at 17:59 +0200, tux john wrote: > Hi. I would like to blacklist a few callers Example: callers with CallerID 0123456789, 9876543210 and 7410258963 are sent to tt-monkeys. Callers from area code 555 are also blocked. In "extensions.conf" file add #include "blacklist.conf" In "blacklist.conf" exten => s/0123456789,1,playback(tt-monkeys) exten => s/9876543210,1,playback(tt-monkeys) exten => s/7410258963,1,playback(tt-monkeys) exten => s/_555XXX,1,playback(tt-monkeys) ... .. . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload
On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote: > bindaddr = all Try: bindaddr=0.0.0.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rasberry pi
I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with Ubuntu Server 14.04. Works fine! :-) Frank On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > SOHO, maybe three hardphones, rasberry pi would suffice? I would be > amazed, but, if so, great. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Including doesn't have any effect
On Mon, 2016-06-06 at 08:08 -0700, Steve Edwards wrote: > The purpose of a subroutine (code that is entered by a gosub and exited by > a return) is to allow the creation of easily reusable code. [snip] Steve Thank you very, very much for your answer. I really appreciated your interesting and detailed explanation. I'll go over the books again and rewrite the little "black box" taking in consideration your suggestions. Thanks again! Best regards Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Including doesn't have any effect
On Mon, 2016-06-06 at 17:47 +0100, Julian Beach wrote:
> exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check
> whether caller blacklisted
As far as I know, Asterisk's database/blacklist function only supports
exact match of caller ID.
If you want to block a specific area code or a block of numbers (eg.
321-654-8XXX) the blacklist function is useless.
Correct me if I'm wrong.
Frank
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Re: [asterisk-users] Including doesn't have any effect
On Sat, 2016-06-04 at 15:19 -0700, Steve Edwards wrote: > Using a 'goto' to exit from a gosub is a bad idea. Why? > A better idea would be > to set a channel variable and check it's value after the return, in the > calling context. The idea is to update the blacklist.conf whenever I want to add or remove a specific number or an entire area code and leave the extensions.conf untouched and to avoid complex regular expressions. > Also, can a 'goto' in a subroutine reference an extension in the calling > context? Seems weird, but 'dialplan' is a weird language :) Well... I'm not an expert and my approach is by "trial and error". It works perfectly. :-) Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Including doesn't have any effect
Another possible approach to blacklist two (or more) specific callers (098765432 and 012345678 as example) In extension.conf #include "blaklist.conf" exten => _+x.,1,Gosub(blacklist,s,1) exten => _+x.,n, exten => black,1,playback(tt-monkeys) In blacklist.conf exten => s/098765432,1,Goto(black,1) exten => s/012345678,1,Goto(black,1) exten => s,1,Return() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set outgoing sip callid ?
In sip.conf [devicename] callerid="Jon Doe" <+123456789> or in extensions.conf exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) exten => 1234,n,Dial(SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipRaider is true for FREE calls?
On Mon, 2016-05-09 at 19:43 -0300, Vitor Mazuco wrote: > VoipRaider the site, says calls to landlines in Brazil... I hope I'm not infringing any mailing list rule by recommending you to take a look to the following providers. I use them with my Asterisk, the rates are good and they allow multiple calls. callwithus.com freelycall.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipRaider is true for FREE calls?
VoipRaider is a service from DELLMONT SARL. This company offers voip services under dozens of different domains (voipcheap, voipdiscount, onevoip,...) Search "Dellmont Sarl" in Google and read the user's reviews. Personally, I would never send a penny to them... Franky -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements
Just a few ideas... 1. Disable all mobile carrier's voicemail and configure a voicemail on your Asterisk. Let Asterisk handle the unanswered calls. 2. If your SIP provider allows multiple calls at the same time, configure Asterisk to call all your SIMs at once (instead of calling the first, wait... calling the second... wait and so on). 3. If your mobile carrier blocks SIP on your data plan, simply configure Asterisk <-> SIP client on your mobile phone to use another port. Or, even better, you can use IAX instead of SIP. On your mobile device install a client that supports IAX (for example, Zoiper). Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?
On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote: > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but > my Huawei E153 is not working in my Asterisk. > But not successes. A little more information from you would be helpful to identify the problem. I have a Huawei USB 3G-stick and it works fine on Asterisk 11. Take a look here: http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/comment-page-1/ Not all Huawei USB modems work out of the box, on some of them voice calling capability has to be enabled first, some need to be upgraded with the latest firmware. Details on this can be found on the original chan_dongle wiki. https://github.com/bg111/asterisk-chan-dongle/wiki/Preparation Before inserting the SIM into your modem please deactivate the PIN on your card. This can be done with any phone. Insert the SIM into your phone, deactivate PIN and you’re done. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handle a call if one phone of a ring group is busy
On Sun, 2016-02-28 at 01:43 +0100, Frank wrote:
> Question: How to give a "busy signal" back to the caller if one
> extension of a ring group is in use? Or redirect the call to voice mail?
Found a solution! :-)
exten => 7654321,1,GotoIf($["${DEVICE_STATE(SIP/111)}"="INUSE"]?Busy,1)
exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/222)}"="INUSE"]?Busy,1)
exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/333)}"="INUSE"]?Busy,1)
exten => 7654321,n,Dial(SIP/111/222/333)
exten => Busy,1,BUSY(10)
exten => Busy,n,Hangup
--
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