Re: [asterisk-users] Disabling CDR for all dialed parties in Asterisk 12
NoCDR doesn't seem to work differently from CDR_PROP(disable), but it is deprecated in Astersisk 12. Anyway, thanks. Please excuse me if this reply goes to wrong place and/or if formatting is incorrect. This is my first time posting/replying in a mailing list. W dniu 2014-09-16 18:54:59 użytkownik Nick Olsen n...@flhsi.com napisał: Not sure if it'll work for your specific use. But I always use app nocdr. exten = 1,1,NoCDR exten = 1,2,Dial(SIP/test,30) Nick Olsen Network Operations (855) FLSPEED x106 From: janusz_1942 janusz_1...@op.pl Sent: Tuesday, September 16, 2014 12:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Disabling CDR for all dialed parties in Asterisk 12 Hello, is it possible to disable the CDR record creation for all dialed parties? From my limited testing it looks like CDR_PROP(disable) is effective only for the first party (the one specified before the first ampersand in the Dial application argument) and I can't find any way to disable it for the other ones (I think the CDR in question is written after the Dial completes). Is it by design? Is there any other known way? Here is my simple testing dialplan: [test] exten = a,1,Set(CDR_PROP(disable)=true) same = n,Dial(Local/chan1@locLocal/chan2@loc) same = n,Hangup() [loc] exten = chan1,1,Set(CDR_PROP(disable)=true) same = n,Answer() same = n,Wait(2) same = n,Hangup() exten = chan2,1,Set(CDR_PROP(disable)=true) same = n,Answer() same = n,Wait(3) same = n,Hangup() --- The following is the result of select * from cdr; on my sqlite cdr backend: --- 1|2014-09-16 18:18:57|test test|test|SIP/test-|Local/chan2@loc-0001;1|Dial|Local/chan1@locLocal/chan2@loc|3|3|ANSWERED|DOCUMENTATION||1410884337.0|| --- Thank you in advance. Best regards, Janusz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mixmonitor - convert wav to mp3/aac
hi, i want convert mixmonitor recorded speech audio from wav to mp3 or aac can you recommend your settings for speech audio? filters, noise elimination, compression ratio, ... i will probably use lame thank you -- --- Marek Cervenka === -- _ -- 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] mixmonitor - convert wav to mp3/aac
Am 18.09.2014 11:06, schrieb Marek Cervenka:
hi,
i want convert mixmonitor recorded speech audio from wav to mp3 or aac
can you recommend your settings for speech audio? filters, noise
elimination, compression ratio, ...
i will probably use lame
Give sox with compiled mp3-support a try:
/usr/bin/sox ${src_file} ${dst_file} lowpass 4000 compand 0.02,0.05
-60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 -7 0.05
I found it on another website ... but I can't remember. Works fine for me.
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[asterisk-users] Asterisk 11.9.0 PRI no ring indications
Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all inbound and outbound calls over the PRI give no ring indications. This is my second converted system and the first system doesn't have this issue. The system is out of the Detroit, MI area and the provider is TDS. One thing of note: I originally started off with 11.12.0, but was having problems with paging and the sound card, so I back dated to 11.9.0 (Same as my first convert) after I deleted the /var/lib/asterisk/modules folder. Machine setup below: lsb_release -a No LSB modules are available. Distributor ID: Debian Description:Debian GNU/Linux 7.6 (wheezy) Release:7.6 Codename: wheezy uname -a Linux livasterisk 3.9.11-custom-3.9.11 #1 SMP Thu Jan 9 12:18:01 EST 2014 x86_64 GNU/Linux dahdi config: cat system.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us echocanceller=oslec,1-23 span=2,0,0,esf,b8zs fxsks=25-32 fxoks=33-48 defaultzone=us loadzone=us chan_dahdi: [channels] ; ; switchtype=national context=pri signalling=pri_cpe echocancel=yes echotraining = yes ;echocancelwhenbridged = yes pridialplan=unknown group=1 rxgain=-1.0 txgain=-4.0 usecallerid=yes callerid=asreceived channel=1-23 Any suggestions would be appreciated! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications
Have you checked your /etc/asterisk/indications.conf file? I know I had a couple systems where I tried to be minimalist with the config files I used, and forgot to bring the indications.conf over from the samples -- the symptom those times was that the caller (inbound, outbound, or extension to extension) wouldn't hear any ringing tone while the call was ringing at the other end. At the CLI, you can use indication show to list all loaded indication types, and indication show zone to see the details about a specific one. I can't remember/find the way to display in CLI what the currently loaded default indication zone is, though there should be a line in the indications.conf file at a minimum. Thank you, Noah Engelberth System Administration MetaLINK Technologies nengelbe...@team-meta.net 419-990-0342 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, September 18, 2014 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 11.9.0 PRI no ring indications Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all inbound and outbound calls over the PRI give no ring indications. This is my second converted system and the first system doesn't have this issue. The system is out of the Detroit, MI area and the provider is TDS. One thing of note: I originally started off with 11.12.0, but was having problems with paging and the sound card, so I back dated to 11.9.0 (Same as my first convert) after I deleted the /var/lib/asterisk/modules folder. Machine setup below: lsb_release -a No LSB modules are available. Distributor ID: Debian Description:Debian GNU/Linux 7.6 (wheezy) Release:7.6 Codename: wheezy uname -a Linux livasterisk 3.9.11-custom-3.9.11 #1 SMP Thu Jan 9 12:18:01 EST 2014 x86_64 GNU/Linux dahdi config: cat system.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us echocanceller=oslec,1-23 span=2,0,0,esf,b8zs fxsks=25-32 fxoks=33-48 defaultzone=us loadzone=us chan_dahdi: [channels] ; ; switchtype=national context=pri signalling=pri_cpe echocancel=yes echotraining = yes ;echocancelwhenbridged = yes pridialplan=unknown group=1 rxgain=-1.0 txgain=-4.0 usecallerid=yes callerid=asreceived channel=1-23 Any suggestions would be appreciated! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice-Recognition / ASR / with barge in
AFAIK, the generic asterisk speech API has an application : - SpeechBackground Not sure if it would work with your custom speech engine, you might have to look at the Generic Speech API to make it work for ur engine. Mitul On Thursday, September 18, 2014, Thorsten Göllner t...@ovm-group.com wrote: Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play a beep. And then we record his voice and realize voice recognition with ispeech (it is an online service). Best regards -Thorsten- -- _ -- 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 -- Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967196 Cell: +91-9820332422 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice-Recognition / ASR / with barge in
Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play a beep. And then we record his voice and realize voice recognition with ispeech (it is an online service). Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications
Have you checked your /etc/asterisk/indications.conf file? I know I had a couple systems where I tried to be minimalist with the config files I used I copied the indications.conf file from the working system, it didn't work. The size was different, I did a reload at the console, but don't know if a restart of asterisk or dahdi needs to be done for it work be read. I'm also planning, after hours, to move back to 11.12.0 with a make config and start again. Maybe moving back to 11.9 wasn't a good idea. Thanks for your input, Doug -- _ -- 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] Polycom DND + Intercom/Paging Override?
On Wed, Sep 17, 2014 at 10:06 PM, Nathan Anderson nath...@fsr.com wrote: BUT Polycom handsets cannot be configured to just listen to RTP being multicasted to a particular multicast IP like many other IP phones can...the signalling for Polycom multicast paging and PTT functionality is completely proprietary and not SIP-based, and in fact the audio itself is not RTP. It is a proprietary audio packet format that has a header prefixed to it containing signalling information, on every audio packet/frame. Therefore nothing else can initiate a multicast page except another Polycom phone on the same layer 2 broadcast domain...you cannot programmatically have Asterisk/FreePBX do this. There is one product that I know of that is Compatible with Polycom paging. The Algo 8180 Audio Alerter. http://www.algosolutions.com/products/Audible-and-Visual-Alerting/8180-sip-audio-alerter.html You can call it via SIP from asterisk and it can multicast in the special Polycom format to your phones. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -- _ -- 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] Polycom DND + Intercom/Paging Override?
- Original Message - Tim, I THINK but I'm not sure that you can do this with the Polycom multicast page function. Have you attempted this yet? Thanks david Given the odd nature of multicast paging with Polycom, I was hoping to avoid such a setup. My recollection is having this work previously with an older version of Asterisk (1.4.x?), and the same handsets. Time to check archived backups... Thank you for the suggestion though, I may have to go that route. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.6-cert6, 11.12.1, 12.5.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 11.6 and Asterisk 11 and 12. The available security releases are released as versions 11.6-cert6, 11.12.1, and 12.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases Please note that the release of these versions resolves the following security vulnerability: * AST-2014-010: Remote Crash when Handling Out of Call Message in Certain Dialplan Configurations Additionally, the release of Asterisk 12.5.1 resolves the following security vulnerability: * AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests Note that the crash described in AST-2014-010 can be worked around through dialplan configuration. Given the likelihood of the issue, an advisory was deemed to be warranted. For more information about the details of these vulnerabilities, please read security advisories AST-2014-009 and AST-2014-010, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert6 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1 The security advisories are available at:  * http://downloads.asterisk.org/pub/security/AST-2014-009.pdf  * http://downloads.asterisk.org/pub/security/AST-2014-010.pdf Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2014-009: Remote crash based on malformed SIP subscription requests
Asterisk Project Security Advisory - AST-2014-009 ProductAsterisk SummaryRemote crash based on malformed SIP subscription requests Nature of Advisory Remotely triggered crash of Asterisk SusceptibilityRemote authenticated sessions Severity Major Exploits KnownNo Reported On 30 July, 2014 Reported By Mark Michelson Posted On 18 September, 2014 Last Updated OnSeptember 18, 2014 Advisory Contact Mark Michelson mmichelson AT digium DOT com CVE Name Pending Description It is possible to trigger a crash in Asterisk by sending a SIP SUBSCRIBE request with unexpected mixes of headers for a given event package. The crash occurs because Asterisk allocates data of one type at one layer and then interprets the data as a separate type at a different layer. The crash requires that the SUBSCRIBE be sent from a configured endpoint, and the SUBSCRIBE must pass any authentication that has been configured. Note that this crash is Asterisk's PJSIP-based res_pjsip_pubsub module and not in the old chan_sip module. Resolution Type-safety has been built into the pubsub API where it previously was absent. A test has been added to the testsuite that previously would have triggered the crash. Affected Versions Product Release Series Asterisk Open Source 1.8.x Unaffected Asterisk Open Source 11.xUnaffected Asterisk Open Source 12.x12.1.0 and up Certified Asterisk 1.8.15 Unaffected Certified Asterisk 11.6Unaffected Corrected In Product Release Asterisk Open Source12.5.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2014-009-12.diff Asterisk 12 Links https://issues.asterisk.org/jira/browse/ASTERISK-24136 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2014-009.pdf and http://downloads.digium.com/pub/security/AST-2014-009.html Revision History DateEditor Revisions Made 19 August, 2014 Mark Michelson Initial version of document Asterisk Project Security Advisory - AST-2014-009 Copyright (c) 2014 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2014-010: Remote crash when handling out of call message in certain dialplan configurations
Asterisk Project Security Advisory - AST-2014-010 ProductAsterisk SummaryRemote crash when handling out of call message in certain dialplan configurations Nature of Advisory Remotely triggered crash of Asterisk SusceptibilityRemote authenticated sessions Severity Minor Exploits KnownNo Reported On 05 September 2014 Reported By Philippe Lindheimer Posted On 18 September 2014 Last Updated OnSeptember 18, 2014 Advisory Contact Matt Jordan mjordan AT digium DOT com CVE Name Pending Description When an out of call message - delivered by either the SIP or PJSIP channel driver or the XMPP stack - is handled in Asterisk, a crash can occur if the channel servicing the message is sent into the ReceiveFax dialplan application while using the res_fax_spandsp module. Note that this crash does not occur when using the res_fax_digium module. While this crash technically occurs due to a configuration issue, as attempting to receive a fax from a channel driver that only contains textual information will never succeed, the likelihood of having it occur is sufficiently high as to warrant this advisory. Resolution The fax family of applications have been updated to handle the Message channel driver correctly. Users using the fax family of applications along with the out of call text messaging features are encouraged to upgrade their versions of Asterisk to the versions specified in this security advisory. Additionally, users of Asterisk are encouraged to use a separate dialplan context to process text messages. This avoids issues where the Message channel driver is passed to dialplan applications that assume a media stream is available. Note that the various channel drivers and stacks provide such an option; an example being the SIP channel driver's outofcall_message_context option. Affected Versions Product Release Series Asterisk Open Source 11.xAll versions Asterisk Open Source 12.xAll versions Certified Asterisk 11.6All versions Corrected In Product Release Asterisk Open Source11.12.1, 12.5.1 Certified Asterisk 11.6-cert6 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2014-010-11.diff Asterisk 11 http://downloads.asterisk.org/pub/security/AST-2014-010-12.diff Asterisk 12 http://downloads.asterisk.org/pub/security/AST-2014-010-11.6.diff Certified Asterisk 11.6 Links https://issues.asterisk.org/jira/browse/ASTERISK-24301 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security
Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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[asterisk-users] conversation record prematurely
I have following line in a context:
...
exten =
_587NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _587NXX,n,MixMonitor(${recordfilename},b)
...
It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.
--
Joseph
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
Your question demonstrates a fundamental lack of Asterisk concepts and
knowledge. You should start by reading http://www.asteriskdocs.org/ and go
from there.Asterisk is not something you can learn in a few days.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach
jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julian
mailto:jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
Thanks Eric, for respectfully pointing that link, it is the reason why I am
posting my question for lack of knowledge. I had been working on Asterisk
for the last 4 years, I am always learning something knew.
- Motty
On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling ewiel...@nyigc.com wrote:
Your question demonstrates a fundamental lack of Asterisk concepts and
knowledge. You should start by reading http://www.asteriskdocs.org/ and
go from there.Asterisk is not something you can learn in a few days.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
*Sent:* Thursday, September 18, 2014 4:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
country - security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk
wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
It is unfortunate
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6
is not helpful to you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 5:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
Thanks Eric, for respectfully pointing that link, it is the reason why I am
posting my question for lack of knowledge. I had been working on Asterisk for
the last 4 years, I am always learning something knew.
- Motty
On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote:
Your question demonstrates a fundamental lack of Asterisk concepts and
knowledge. You should start by reading http://www.asteriskdocs.org/ and go
from there.Asterisk is not something you can learn in a few days.
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach
jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julian
mailto:jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk
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[asterisk-users] Record call ends in 10min
In my context I have:
exten = _NXX,1,Set(CHANNEL(musicclass)=default)
exten =
_NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _NXX,n,MixMonitor(${recordfilename},b)
but the recorded conversation ended in 10min so it = 600sec
I was looking in asterisk configuration file for 600 pertaining recording but
I couldn't fine any.
--
Joseph
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
absolutely not what I meant, I really meant to say thank you for
respectfully pointing that out.
-Motty
On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling ewiel...@nyigc.com wrote:
It is unfortunate
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6
is not helpful to you.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
*Sent:* Thursday, September 18, 2014 5:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
country - security mechanism
Thanks Eric, for respectfully pointing that link, it is the reason why I
am posting my question for lack of knowledge. I had been working on
Asterisk for the last 4 years, I am always learning something knew.
- Motty
On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling ewiel...@nyigc.com wrote:
Your question demonstrates a fundamental lack of Asterisk concepts and
knowledge. You should start by reading http://www.asteriskdocs.org/ and
go from there.Asterisk is not something you can learn in a few days.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz
*Sent:* Thursday, September 18, 2014 4:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud
country - security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk
wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
--
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Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
My apologies, I misunderstood. I’m glad the link was helpful.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 5:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
absolutely not what I meant, I really meant to say thank you for respectfully
pointing that out.
-Motty
On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote:
It is unfortunate
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6
is not helpful to you.
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 5:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
Thanks Eric, for respectfully pointing that link, it is the reason why I am
posting my question for lack of knowledge. I had been working on Asterisk for
the last 4 years, I am always learning something knew.
- Motty
On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote:
Your question demonstrates a fundamental lack of Asterisk concepts and
knowledge. You should start by reading http://www.asteriskdocs.org/ and go
from there.Asterisk is not something you can learn in a few days.
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of motty cruz
Sent: Thursday, September 18, 2014 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country
- security mechanism
Thank you Julian,
would it be possible to block calls to international calls except certain
countries? I just want to make sure that if attackers try to place calls
outside the states they not succeed.
Thanks,
Motty
On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach
jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk wrote:
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julian
mailto:jb_s...@trink.co.ukmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] Record call ends in 10min
On 09/18/14 15:43, Joseph wrote:
In my context I have:
exten = _NXX,1,Set(CHANNEL(musicclass)=default)
exten =
_NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _NXX,n,MixMonitor(${recordfilename},b)
but the recorded conversation ended in 10min so it = 600sec
I was looking in asterisk configuration file for 600 pertaining recording but
I couldn't fine any.
I don't have any maxduration set in any config file.
maxduration: maximum recording duration in seconds. If missing or 0, there is
no maximum.
--
Joseph
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Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications
Ringback problems are a pain in the neck to troubleshoot. You don't mention your endpoint, but if the endpoint is sip, play around with the prematuremedia and progressinband options in sip.conf.The comments for these two settings in sip.conf.sample are completely and totally confuzing. Try different compications and see if any of them make any difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, September 18, 2014 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications Doug Lytle wrote: I'm also planning, after hours, to move back to 11.12.0 with a make config and start again. Maybe moving back to 11.9 wasn't a good idea. Well that didn't work. Even started with a fresh set of configuration files. A basic chan_dahdi.conf and a basic dahdi system.conf. Audio passes fine, I can even specify a default music on hold to play during the dial (Thinking of playing a ringing sound). I'm just not getting any ringing. Is it possible the card is bad? It's a Digium, Inc. Wildcard TE220 dual-span T1/E1/J1 card 3.3V (PCI-Express) (5th gen) (rev 02) Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conversation record prematurely
On Thu, Sep 18, 2014 at 3:16 PM, Joseph syscon...@gmail.com wrote:
I have following line in a context:
...
exten =
_587NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _587NXX,n,MixMonitor(${recordfilename},b)
...
It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.
Without further information the only reason I could see would be the
'b' option in use for MixMonitor. If the channels were no longer
bridged it would stop recording. That is according to the
documentation.. which every once in a while is wrong. Other than that,
it should record as long as the channel is bridged.
Can you pastebin a log showing that particular call?[1]
[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Record call ends in 10min
On Thu, Sep 18, 2014 at 5:00 PM, Joseph syscon...@gmail.com wrote: On 09/18/14 15:43, Joseph wrote: I don't have any maxduration set in any config file. maxduration: maximum recording duration in seconds. If missing or 0, there is no maximum. Joseph, please don't start new threads on a duplicate topic so quickly. It can create noise in the list and cause confusion when some people reply to your second thread vs your first. I've replied to your first thread. Re: maxduration, I believe that is an option for Record and not MixMonitor, but I could be wrong. Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications
Eric Wieling wrote: You don't mention your endpoint Both ends of the PRI. In house the end points are SIP phones, but calling from a sip phone (Polycom) to our remote office, there is no ringing. I'll be on site again this Saturday. I may end up putting the old 1.4x box back into place, I did get it working again. Doug -- _ -- 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] Record call ends in 10min
On 09/18/14 16:00, Joseph wrote:
On 09/18/14 15:43, Joseph wrote:
In my context I have:
exten = _NXX,1,Set(CHANNEL(musicclass)=default)
exten =
_NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _NXX,n,MixMonitor(${recordfilename},b)
but the recorded conversation ended in 10min so it = 600sec
I was looking in asterisk configuration file for 600 pertaining recording but
I couldn't fine any.
I don't have any maxduration set in any config file.
maxduration: maximum recording duration in seconds. If missing or 0, there is
no maximum.
Apology about it.
I'll try MixMonitor without b
exten = _NXX,n,MixMonitor(${recordfilename})
--
Joseph
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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?
On Thursday, September 18, 2014 10:31 AM, John Kiniston wrote: There is one product that I know of that is Compatible with Polycom paging. The Algo 8180 Audio Alerter. [snip] You can call it via SIP from asterisk and it can multicast in the special Polycom format to your phones. Wow, I had no idea! I have looked at SIP-based PAs in the past, including this one, but this completely escaped my attention. I just browsed through the manual, and sure enough, this is an advertised feature. Kinda weird that you have to buy an all-in-one loudspeaker to acquire a device that can act as a SIP-to-Polycom-multicast bridge...it would be nice if they sold a cheaper version that omitted the speaker. (Or, even better yet, if Asterisk just supported this natively so that you didn't have to buy some hardware box.) But still, it's nice to know that this exists and is an option. Thanks for the heads-up! -- Nathan Anderson First Step Internet, LLC nath...@fsr.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
