Re: [asterisk-users] Block Specific Number on Inbound
Here's what I do... Changed some variables for obscurity. 911 is the inbound
#... exten 6000 rings to SIP/TEST
exten = 911,1,GotoIf(${BLACKLIST()}?blacklisted)
exten = 911,n,Macro(stdexten,6000,SIP/test)
exten = 911,n,Playback(transfer,skip)
exten = 911,n(blacklisted),Goto(blacklisted,s,1)
Blacklisted context uses zapateller and plays the intercept message
[blacklisted]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Zapateller
exten = s,4,Zapateller
exten = s,5,Playback(ss-noservice)
exten = s,6,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Sheldon
Sent: Thursday, December 29, 2011 9:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Block Specific Number on Inbound
Check out the X Boy/Girl friend feature.
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
Around the middle of the page.
Stu
-Original Message-
From: Kevin Oravits korav...@rcolegal.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound
Date: Fri, 30 Dec 2011 01:39:46 +
Greetings,
Is there a way to block a specific inbound number? I’ve found code online for
blocking all nocallerid and all 800, etc. but nothing for a specific number. My
company is wanting me to block a specific number. Is this possible in Asterisk
1.4 and 1.6 or do I need to go through my Service Provider?
Thanks,
Kevin Oravits
Phone Sys Admin
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Re: [asterisk-users] Block Specific Number on Inbound
Take a look at Blacklist I love that command and love to send nice intercept messages to the other side J From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits Sent: Thursday, December 29, 2011 8:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Block Specific Number on Inbound Greetings, Is there a way to block a specific inbound number? I've found code online for blocking all nocallerid and all 800, etc. but nothing for a specific number. My company is wanting me to block a specific number. Is this possible in Asterisk 1.4 and 1.6 or do I need to go through my Service Provider? Thanks, Kevin Oravits Phone Sys Admin -- _ -- 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] Cisco AS5400XM
Also used for calling card platforms :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas Sikkema Sent: Thursday, October 06, 2011 5:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco AS5400XM On 10/6/11 11:25 PM, Kyle Sexton wrote: I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP signaling. Has anyone had any experience with these devices? The feature cards that Cisco sells can be a little confusing. I'm thinking something like below is what I need. (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem) (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card Any thoughts would be appreciated. Thanks. I've used them in the past and still use the little brother (AS5350XM). I have no experience with T1s, but I used them to convert EuroISDN E1s to SIP. They were very stable (I don't think I've ever seen one crash) but can be a pain when you want to set them up. These machines were originally designed as modembanks for internet access so the default config has an interface for every B channel. That is a pain to browse through the configuration. Grouping them solves this. Make sure you understand how to route calls using dialpeers, and make sure you understand this before putting them in service. These are very, very capable machines with lots of useful configuration options. Make sure you buy enough DSP channels to cover all simultaneous calls that need transcoding, we generally bought enough DSP cards so we could transcode all simultaneous calls. If you add it all up we were actually buying more DSP channels than E1 channels were available, for some reason Cisco designed the machine like this, perhaps to cover for slow call teardowns occupying DSPs too long. -- Andreas Sikkema -- _ -- 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] VoIP Abuse to Twitter (real time VoIP Abuse)
Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.net
wrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=get
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
search=0x2BF7D83F210A95AF
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
I personally would never install a GUI o/s. By doing so you always open yourself up to more security concerns.. Packages / ports / etc. Course one might argue - it's behind a firewall.. In my professional experience with running numerous ISP and VoITSPs the rule has always been install the minimum needed software to accomplish the goal. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy Sent: Monday, September 12, 2011 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ? I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. I look forward to your input. Thanks -- _ -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
Well you are correct - I did not include a discussion on performance impacts including disk I/O etc. It is true that by installing a GUI o/s additional init.d (startup) services will fire.. Additional libraries will be inclusive etc. This is why I say minimal is always better. Also take for example risk mitigation with security aspects. If you minimize the number of libraries (think windows DLL's) you have installed you also thus minimize your potential exposure. Again - this is just my recommendation and experience. Firewalls are great at blocking things and in theory - sure you could nmap your box and look for open ports and conceal them. I remember a Solaris engineer we had once - he bragged and bragged about his qualifications on Sun Solaris. Just to find out that he installed a bunch of GUI tools just so that he could install Oracle drivers. Further he didn't remove or lock down that exposure. Start minimal and work your way up. Now for my poke / razz - GUI's in server grade operating systems have made people a little to reliant on them. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy Sent: Monday, September 12, 2011 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ? On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.com wrote: I personally would never install a GUI o/s. By doing so you always open yourself up to more security concerns.. Packages / ports / etc. Course one might argue - it's behind a firewall.. In my professional experience with running numerous ISP and VoITSPs the rule has always been install the minimum needed software to accomplish the goal. Thanks for the reply. I was worried the list would find it a trite and irritating question. I was expecting someone to tell me that even with the GUI component running in the background, the graphical processes have the potential to mess up the streams. I guess I should confess that I'm always a bit surprised to remember that asterisk doesn't require a real time OS ! Have you really exposed much more if you install the GUI components and normally run at init 3 ? Thanks again. -- _ -- 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] Reporting for Asterisk Call Center
www.buildityourself.org :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Friday, September 09, 2011 2:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reporting for Asterisk Call Center Hi All; Anyone advise for a free (open source) reporting to be used for asterisk call center? Regards Bilal -- _ -- 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] Prompt for PIN After dialing
Have you looked at pin sets in freepbx / trixbox / elastix? I haven't tested it myself - but I know the feature is present there -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, September 02, 2011 11:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prompt for PIN After dialing Hello All, We would like to change our dialplan a bit so that after a user dials a number (any number, including domestic, international, internal) Asterisk firsts prompts the user for a PIN before actually allowing the call to go through. I know I could setup an IVR that would accomplish this but I'd prefer not to have the users first call an internal extension before they dial out. I want them to be able to dial the destination number directly, have asterisk intercept and prompt for password, then either allow the call or play a .gsm file and hangup if the PIN is incorrect. We are using AELs, and not the exten,x,x format. Thanks in advance, Brandon -- _ -- 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] Avaya to Asterisk Voice mail
Search the forum - I believe I remember a recent exchange on this subject From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dustin fails Sent: Tuesday, August 30, 2011 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Avaya to Asterisk Voice mail Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue line. The issue I am having is Avaya is sending the originating caller id not the station id so Asterisk see that originating id so I can't route the call correctly in Asterisk. Thanks! Dustin -- _ -- 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] 1.8.5 Voicemail duration incorrect
https://issues.asterisk.org/jira/browse/ASTERISK-16981 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Thursday, August 25, 2011 3:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.8.5 Voicemail duration incorrect Hi, Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston: Anyone else seen this? I saw a jira but was in feedback status.. I just checked with a voicemail of 60 seconds. It was reported in .txt-file with a duration of 19 seconds. So there is a bug. Do You have a link to the Jira issue? Karsten -- _ -- 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] 1.8.5 Voicemail duration incorrect
Anyone else seen this? I saw a jira but was in feedback status.. -- _ -- 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] Assistance sending mass sms to cellphones
This is off topic... Asterisk will not provide you with the ability to SMS random cell phones. Being able to transport the SMS yourself is a grewling process.. Look at software like Kamel... Basically you have three options: ( a ) cheat and use the email method - i.e. determine everyone's carrier and use the email address equivalent ( b ) utilize a 3rd party to transmit the sms for you (cost) and they might end up doing ( a ) above without you knowing ( c ) spend lots of money and headaches transporting sms yourself. Either way it's off-topic and not related to Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Friday, August 05, 2011 11:42 AM To: asterisk Subject: [asterisk-users] Assistance sending mass sms to cellphones Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- 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] Assistance sending mass sms to cellphones
Seriously Again? This is off topic... Asterisk will not provide you with the ability to SMS random cell phones. Being able to transport the SMS yourself is a grewling process.. Look at software like Kamel... Basically you have three options: ( a ) cheat and use the email method - i.e. determine everyone's carrier and use the email address equivalent ( b ) utilize a 3rd party to transmit the sms for you (cost) and they might end up doing ( a ) above without you knowing ( c ) spend lots of money and headaches transporting sms yourself. Either way it's off-topic and not related to Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Friday, August 05, 2011 12:42 PM To: asterisk Subject: [asterisk-users] Assistance sending mass sms to cellphones Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- 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] Assistance sending mass sms to cellphones
When you say expensive... You are talking about pennies per SMS... Again - if you want to cheat and go the email route - that would be free... It's unreliable and requires some thought... If you want more information / consulting contact me off-list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Friday, August 05, 2011 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones Robert. Thanks for replying. --- On Fri, 8/5/11, Robert Huddleston rhuddles...@gmail.com wrote: From: Robert Huddleston rhuddles...@gmail.com Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Friday, August 5, 2011, 11:50 AM This is off topic... Asterisk will not provide you with the ability to SMS random cell phones. We actually have a group of people belonging to a rotary club and we wanted to automate the sms process... is not random cell phones. Being able to transport the SMS yourself is a grewling process.. Look at software like Kamel... Basically you have three options: ( a ) cheat and use the email method - i.e. determine everyone's carrier and use the email address equivalent ( b ) utilize a 3rd party to transmit the sms for you (cost) and they might Looks like this is the easiest option but, very expensive for what we really want to do. end up doing ( a ) above without you knowing ( c ) spend lots of money and headaches transporting sms yourself. Either way it's off-topic and not related to Asterisk. Sorry, didn't think this wasnt an asterisk related question. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Friday, August 05, 2011 11:42 AM To: asterisk Subject: [asterisk-users] Assistance sending mass sms to cellphones Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- 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 -- _ -- 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] T38 Fax
Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -- _ -- 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] T38 Fax with Grandstream HT-502
My apologies - yes.. Grandstream HT-502... Apparently finding a t.38 provider is also another struggle... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Monday, August 01, 2011 1:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] T38 Fax with Grandstream HT-502 On 08/01/2011 12:02 PM, Robert Huddleston wrote: Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. You'd be more likely to get relevant responses if you had included the information about the HT-502 in your message subject :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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 -- _ -- 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] T38 Fax
Thanks - and did you find a provider with T.38 DIDs? I don't see many pay as you go providers with T.38 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Monday, August 01, 2011 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T38 Fax On 2/08/2011 1:02 AM, Robert Huddleston wrote: Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks Yes, it works. I currently have latest firmware installed and it still works in T.38. I am using UDP transport for this device as I seem to encounter problems with TCP or TLS. I am currently running Asterisk 1.8.5.0. Product Model: HT-502 V1.1C Software Version: Program-- 1.0.5.5Bootloader-- 1.0.0.9Core-- 1.0.5.2Base-- 1.0.5.2 Some settings I have set and you may wish to check for the FXS port are; Force INVITE: (X) No ( ) Yes (Always refresh with INVITE instead of UPDATE) Send Re-INVITE After Fax: ( ) No (X) Yes VAD: ( ) No (X) Yes Symmetric RTP: (X) No ( ) Yes Fax mode: (X) T.38 (Auto Detect) ( ) Pass-Through Fax tone detection mode: ( ) Caller (X) Callee ( ) Caller or Callee Jitter buffer type: (X) Fixed ( ) Adaptive Jitter buffer length: (X) Low ( ) Medium ( ) High You will need to ensure you are using redundancy mode instead of FEC. I am able to send a fax via my voice provider seemingly without errors even though ECM is not enabled, this is because redundancy mode is working as expected on the outbound communication. Unfortunately my voice provider only sends one data item in the incoming UDPTL hence the occasional missed line. Here is an extract from my sip.conf [general] . . t38pt_udptl=yes,redundancy,maxdatagram=400 . . [906] ; Grandstream HT502 FXS Port ; Analogue FAX Modem attached type=friend defaultuser=906 md5secret=c5bca943c9b0cc303c496fbf9d48a48e call-limit=1 disallow=g722 transport=udp qualify=yes directmedia=no host=dynamic context=FAX-T38 faxdetect=no deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.0.0.0 permit=172.16.0.0/255.240.0.0 permit=192.168.0.0/255.255.0.0 Larry. -- _ -- 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] MoH - conversion command
Personally I like to just hook up an old ghetto blaster / boombox to the
line in port on my sound card :)
Kidding aside - I think audio quality for MoH is not always going to sound
as nice as you might want.
I mostly stream online radio over my MoH and the quality is not the
greatest.
Maybe it's my SIP provider - or maybe just the notion of streaming audio
from an internet stream.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, July 28, 2011 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MoH - conversion command
On Thu, 28 Jul 2011, Mike wrote:
I?ve got a hold of Royalty-free Classical music (a safe choice for
most of my customers) and I`ve been trying to convert them to the
normal telephony/Asterisk format using sox. Unfortunately, it sounds
really bad.
I convert files using:
sox ${INPUT} -c 1 -s -w -r 8000 /tmp/$$.wav
What does your sox command line look like?
Can you post a link to 'before' and 'after' files?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Stun Server
I like Xen. It's free and rock solid. VMWare is great but their money greedy. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, July 27, 2011 9:30 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Stun Server We have been running a windows stun server for 5 years now and I would like to change to either a linux of freebsd based unit to phase out the old XP box in our datacenter. What should I look at that would be a good replacement. The windows box has worked but the hardware is at end of life and I want to move it to a vm and I don't want Windows. Any advise is apperciated. Thanks zktech -- _ -- 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] Lightning and thunder
Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called rebootflag. set it to 1 after you issue a shutdown -r -n now. then check it in init.d script. Pseudo code In init.d / startup scripts If /etc/manualreboot = 0 or file not found echo 1 /etc/manualreboot /sbin/shutdown -r -n now end if From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claude Hayn Sent: Wednesday, July 27, 2011 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Lightning and thunder We are frequently losing power during lightning storms. (Yes we have UPS, but often by the time power comes back up the UPS has run out of juice) We are using Asterisk with a T1/PRI card as a front end connected to our PBX. Whenever there is a power outage both the Asterisk box and the PBX automatically reboot when power returns. The Asterisk box automatically reconnects to the ITSP SIP Peer, and the PBX to the T1/PRI Card Asterisk box. Incoming calls connect, but outbound calls will not complete until the Asterisk box is manually rebooted again. Does anyone know of a solution for this issue? Having to get up in the late night to manually reboot the Asterisk box is getting old! Thank you, Claude -- _ -- 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] Lightning and thunder
I agree - using powerchute or another ups software clean shutdown is great. My response was a scripted way to resolve the reboot issue based on what the writer asked for. Additionally loop wouldn't happen. That's why I wrote echo 1 some file and if check that file. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, July 27, 2011 10:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Lightning and thunder This is the right idea - have your UPS write power loss shutdown when it has to stop the machine, then check for that when you come back up and reboot when you see it (of course you would need to log something else to prevent a loop of reboots). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Huddleston Sent: Wednesday, July 27, 2011 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Lightning and thunder Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called rebootflag. set it to 1 after you issue a shutdown -r -n now. then check it in init.d script. Pseudo code In init.d / startup scripts If /etc/manualreboot = 0 or file not found echo 1 /etc/manualreboot /sbin/shutdown -r -n now end if From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claude Hayn Sent: Wednesday, July 27, 2011 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Lightning and thunder We are frequently losing power during lightning storms. (Yes we have UPS, but often by the time power comes back up the UPS has run out of juice) We are using Asterisk with a T1/PRI card as a front end connected to our PBX. Whenever there is a power outage both the Asterisk box and the PBX automatically reboot when power returns. The Asterisk box automatically reconnects to the ITSP SIP Peer, and the PBX to the T1/PRI Card Asterisk box. Incoming calls connect, but outbound calls will not complete until the Asterisk box is manually rebooted again. Does anyone know of a solution for this issue? Having to get up in the late night to manually reboot the Asterisk box is getting old! Thank you, Claude -- _ -- 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] NAT yes
Also consider the setting localnet in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Tuesday, July 26, 2011 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] NAT yes On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] My Asterisk Box was hacked
When I get hacked I typically run a rootkit checker http://www.chkrootkit.org/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace Sent: Thursday, July 21, 2011 2:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] My Asterisk Box was hacked On Thu, 21 Jul 2011 13:29:09 +0800 Malvin Rito mr...@mail.altcladding.com.ph wrote: My asterisk box was hacked! Can anyone help on how do I secure my asterisk box, currently my box is installed with 2 NIC. 1st NIC is for LAN access and 2nd NIC has a public IP which is registered to our VoIP Provider. Seven Steps to Better SIP Security with Asterisk http://blogs.digium.com/2009/03/28/sip-security/ -- _ -- 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] Multiple Asterisk Sessions on same machine
I prefer How do we do that? Isn't Asterisk a SIP Proxy ;)? That's a good question... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, July 19, 2011 2:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Asterisk Sessions on same machine On 07/19/2011 01:16 PM, Alex Balashov wrote: On 07/19/2011 02:15 PM, Kevin P. Fleming wrote: Actually, you can do this with one installation of Asterisk, and a separate set of config files and data directories. When the Asterisk executable is started, the '-C' option can be used to point to an asterisk.conf file; that file can then tell it where all the other config files and the data directories are located. If you are using one of the init scripts, then yes, that would need to be duplicated and modified. How, do you suppose, would the complexity of that compare to chrooting two installations? They are probably equal in terms of complexity and effort required; just different methods. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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 -- _ -- 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] Requires
Boy if only it was Enron :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, July 18, 2011 8:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Requires First they came and said that instead of offices, doors and hallways, we should have massive, open-plan seating or grungy, industrial cubicle farms, because open spaces mean open companies! It's safe to say the advice did not fall on deaf ears. Now, we're ready to take openness to the next level. Is asterisk-users ready to be copied on all internal company correspondence? Challenge accepted. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] Requires
Alex you are my role model... Next time I'm in Atlanta - let's do lunch! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, July 18, 2011 9:08 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Requires On 07/18/2011 09:00 AM, Robert Huddleston wrote: Boy if only it was Enron :) Baby steps. Success is not built overnight; you have to work your way up the totem pole of fleecing people. Start small: persistently ask basic, RTFM-grade newbie questions while assigning yourself pompous, self-aggrandising titles like Asterisk Engineer. Keep it up, and you'll be crashing national economies with fraudulently constructed multi-billion dollar securitised debt tranches in no time. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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 binaries on CentOS version 6
I stand amused that people want to experiment with VoIP and Asterisk - but aren't willing to: ( a ) Read wiki / manuals / faqs ( b ) demand packages for their o/s This ain't windows folks :) ./configure make make install Is really simple :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Thursday, July 14, 2011 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk binaries on CentOS version 6 On Thursday 14 Jul 2011, Kaushal Shriyan wrote: Hi, Any time line of availability of Asterisk binaries on CentOS version 6. Yeah . as soon as someone compiles them :) Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even take long anymore (on any target system with the grunt to run Asterisk). The only thing to beware of is, if configure complains that you need a package that you already have, then you need the corresponding -devel package. Go on, live a little! Just because you're using CentOS, doesn't mean you have to be boring ;) -- AJS Answers come *after* questions. -- _ -- 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] CDRs
Read the wiki / manuals From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Tuesday, July 12, 2011 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDRs Hi Like we can define cdr field format for csv, is it possible to define if cdrs are stored in a database? Also, what will be size limit for database CDR storage ? -- _ -- 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] Cisco IP Phones and Skinny in asterisk
If memory serves isn't that support contract include broken phones / parts too? I thought I read that if my phone Is broken - it is covered From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, June 20, 2011 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.com wrote: You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) The support contract from Cisco is only US $8.99 on CDW I really hate to link to my own blog, but I do have a post on there that details how to setup a 79x1 phone using SIP firmware with asterisk. Click the link in my signature and go to the Blog and you should be able to easily find the relevant post. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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] SMS with Asterisk
Hahahah Baltimore and SE DC. How about Philly too J From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, June 21, 2011 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SMS with Asterisk On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby wcse...@selbytech.com wrote: On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Two requests, not from me but the community. 1. Don't top post *cough* 2. When you find your solution, reply to this thread so others will be (silver) spoon fed the answers and blindly accept them without trying things and going through a learning curve and experimentation when they find your post in Google. I hear some people are actually deploying their asterisk solutions in war zones and are taking heavy fire while they're looking for answers - seems like it would make their life a whole lot easier (and safer!) if people posted simple responses on this list when suggestions worked for them... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com LOL at the haters. 1. It was joke for those with senses of humor and know me (Randy got it), but I top post when others do. I bottom post when others do. I just go with the flow. I am not uptight about it. 2. I have never heard that but it may be true. Personally, I have been shot at on top the Iraqi Government building in the IZ from the Red Zone. I was setting up and troubleshooting the Motorola Canopy WiFi system. Just a few 7.62x39 rounds, nothing I would call heavy fire. The only Heavy Fire I took was standing on top of one of the buildings at the FOB trying to trace a cable and the ricochets from the firing range were landing all over the place. That happens when 30 guys are training with AKs and a T-Wall as the backstop. I have deployed Asterisk systems in war zones many times, in West African countries, Iraq, Baltimore and South East DC. I would certainly seek shelter/defensive position if there was gun play. LOL, you can wish yourself into a gun fight but you cannot wish yourself out. It would also be a whole lot easier for someone to physically feed me so my hands could be free to work in hostile environments, maybe an LN can bring me a portable toilet and make sure it is fresh, that would make everything so easy and easy is what we all want. Heck, I could just set it up at the FOB and then deploy it. At any rate, I asked the guy to post his success, so I am not sure why you posted, but thanks. It only takes 10% truth to make a legend. Thanks, Steve T -- _ -- 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] Cisco IP Phones and Skinny in asterisk
I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, June 20, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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
[asterisk-users] Ground Start ATA / VOIP Gateway
Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. -- _ -- 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-biz] Ground Start ATA / VOIP Gateway
I only need 4 fxs. I looked at the IAD2431 but it uses T1/E1 as WAN. If I could assign Fast Ethernet to WAN that would be great. Budget is not that great From: asterisk-biz-boun...@lists.digium.com [mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of Sum Ding Wong Sent: Tuesday, June 14, 2011 3:23 PM To: Commercial and Business-Oriented Asterisk Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-biz] Ground Start ATA / VOIP Gateway Cisco Gateways can do ground start signaling. What is your budget and port density need? On Tue, Jun 14, 2011 at 1:19 PM, Robert Huddleston rhuddles...@gmail.com wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- _ -- 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] Ground Start ATA / VOIP Gateway
Ya - customer is on a nice NEC SV8100.. The card is a ground start card.. they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown cross-connect. But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and want to use Ethernet for wan. So IAD2431 would be great - but if it only allows T1/E1 for WAN - I'm shot. From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Robert Huddleston Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway Robert Huddleston wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. I don't know of any ATA that will do GS An RJ-21 is the designation for a 66 block with 25 pair connector on the side GS is available with many channel banks though a T1 card and channel bank might be overkill for your application. Is this to go into a legacy switch? Most have line cards that can be easily switched to Loop Is this in the US, or ??? John Novack -- Dog is my Co-pilot -- _ -- 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] Ground Start ATA / VOIP Gateway
I'll have to look at that then - as I thought the card actually said Ground Start on it.. I may have missed or it was scratched off the word loop start From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 5:20 PM To: Robert Huddleston Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway The SV8100 can do either ground or loop Assuming you can access the system it can easily be changed. Programming manual here: http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf the original installer may have locked it down, but it CAN be changed. John Novack Robert Huddleston wrote: Ya - customer is on a nice NEC SV8100.. The card is a ground start card.. they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown cross-connect. But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and want to use Ethernet for wan. So IAD2431 would be great - but if it only allows T1/E1 for WAN - I'm shot. From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Robert Huddleston Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway Robert Huddleston wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. I don't know of any ATA that will do GS An RJ-21 is the designation for a 66 block with 25 pair connector on the side GS is available with many channel banks though a T1 card and channel bank might be overkill for your application. Is this to go into a legacy switch? Most have line cards that can be easily switched to Loop Is this in the US, or ??? John Novack -- Dog is my Co-pilot -- Dog is my Co-pilot -- _ -- 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
