Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
On 15-06-15 08:48 PM, Matt Darnell wrote: In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built into the unit. Unfortunately the FXS ports are just ATA's off of Asterisk, no different than a SPA2012 unit. The 550 is getting long in the tooth and very expensive for a few FXS ports, what are you folks doing when someone has a need? It can be a modem for the power company to read the meter, a postage machine that needs to get more postage, an alarm system,etc. Finding less and less need for analog circuits. Alarm - envisalink, postage machine - ethernet based, pos terminal - ethernet. Fax is really the only need recently, and even that has alternatives like emailing scans that most people prefer now. Is the customer buying a POTS line and splitting it the only other way? Thanks, Matt -- _ -- 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 on Windows
On 12/04/2013 10:22 AM, Gregory Malsack wrote: Its beyond disgusting. If it was not for legacy garbage nothing from m$ would be left in my datacenter. Saying you are an expert Linux user is just a joke when you don't understand the poor architectural choices that come with windows and why it can never be a real robust operating system. That's just disgusting If you want to run your phones on WindBlows use lync Should be plenty point and click easy for you On 12/04/2013 09:19 AM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. -- _ -- 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] LUA
On 07/18/2013 09:56 AM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the make linux install command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. you're kidding right ? Why not just plug in the box somewhere else, do your install and move it back ? OS CentOS 6.4 Asterisk version 1.8.13.0 11.4 $ find / -name *lua* /usr/local/include/lua.h /usr/local/include/lua.hpp /usr/local/include/lualib.h /usr/local/include/luaconf.h /usr/local/lib/lua /usr/local/lib/liblua.a /usr/local/bin/luac /usr/local/bin/lua /usr/lib64/liblua-5.1.so /usr/bin/luac /usr/bin/lua Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 -- _ -- 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] PoE module
On 07/14/2013 03:12 PM, bilal ghayyad wrote: check ebay there are lots of wide power injectors with 4 8 or 16 ports. Hello; We have a cisco switches but they are not PoE and we need only to have PoE device so the cables come for it first to provide the power and then goes to the switch (to be like batch panel), is there something like this that can be used for the IP Phones? 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] Looking for a reporter for SQLite3 with Lighttpd and PHP
On 04/09/2013 01:44 PM, Daniel - Asterisk wrote: sqlite is not really a multiuser dbms, so really its hard to build tools that do what you want without causing problems in the way it operates. You'd be better off running postgres or mysql (yuck I said it), and using one of the many tools which exist for those. Bear in mind all the wear and levelling issues with databases if you are using solid state storage only (I wouldn't but its possible to do) Hello List, here I go again, I'm looking for an interface to access my Sqlite3 DB which holds my CDR and SIP realtime tables. I'm running Asterisk 1.8.20.1 on my Raspbian distro (Debian Wheezy). I've tried http://astcdrview.berlios.de/ without success after following installation guide and it seems to be an abandoned project. If you know about a tool or product to download CDR reports and update SIP realtime tables please let me know. Regards, Elder D. Arohuanca dCAP Lima - Peru On Wed, Mar 20, 2013 at 11:58 AM, Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com wrote: Hello everyone, I wonder if there's a product that I can install on my debian-based server to extract CDRs (it'd be better if Excel's downloads are available), also it would be desirable if I can access additional table to update rows (e.g. sip for realtime) Please let me know what you know. Best Regards, Elder D. Arohuanca dCAP Lima - Peru -- _ -- 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 SMS()
On 02/19/2013 11:20 AM, Christopher Harrington wrote: I was always under the impression you needed to either use a cellphone type device to send them using your account, or send on to one of the aggregators who have apis for this. For low volume stuff, you can simply send an email number@providergateway and it will hit the phone network as an sms. On Tue, Feb 19, 2013 at 10:12 AM, Nicholas Johnson nejohns...@me.com mailto:nejohns...@me.com wrote: On Feb 19, 2013, at 10:41 AM, Christopher Harrington wrote: On Tue, Feb 19, 2013 at 9:14 AM, Nicholas Johnson nejohns...@me.com mailto:nejohns...@me.com wrote: All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same = n,SMS(hello,a,17654307001 tel:17654307001,hello nick) Let's start out by figuring out what hardware you have. Is Asterisk connected to the PSTN? What is physically delivering the SMS to the carriers? Also, when I run `core show application SMS` it talks about some software called smsq. Are you running that software? Thanks for the help. Right now I'm running asterisk on a raspberry pi using a phone number from flowroute. Is using a company like flowroute the same as connecting to the PSTN? Also i've tried to install smsq but I couldn't find any good documentation to get it setup properly. So no, I'm not using smsq. I'm not well informed, but it appears that you need to (at a minimum) provide some sort of interface to connect to a hardware interface for this process. Google ETSI ES 201 912. Having looked at Flowroute, they don't appear to mention SMS anywhere on their website, so I am going to go out on a limb and say that they will not provide what you need to send an SMS. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Single = sign and double == sign.What is the difference and when to use the two properly?
On 01/11/2013 12:20 PM, A J Stiles wrote:
I try to write comparisons as != where possible and then there is no
confusion and less mistakes possible.
Most compilers will warn on the example below now.
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false), Hangup()
would this be saying the same thing as above then?
exten = _,answer()
same= n,Set($[${a}=1]?true:false] -- single equal sign
in essence wouldn't i get the same result ? im confused on the double and
single equal sign and when to use the difference of the two. Would i get
the same result in both these expressions?
Generally, one = sign means you're telling. Two == signs means you're asking.
It's amusing (for sadists) to see ex-BASIC programmers trip up over this and
write something like this:
if (denominator = 0) {
printf (Can't divide by zero!\n);
} else {
answer = numerator / denominator
};
This will never print Can't divide by zero! because you are actually
assigning a value to a variable right there in the conditional, and returning
the assigned value. Since this is zero, the if() will fail and drop through
to the else clause -- and then, just to confuse you, the program will crash
with Floating point exception anyway.
--
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Re: [asterisk-users] DIDForSale spam
On 01/09/2013 01:49 PM, Steve Edwards wrote: I was about to reply 'no' but thought to check my spam logs so now I reply 'yes.' I got a few of them actually. -- _ -- 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] IVR platform for a mobile operator
On 01/09/2013 03:52 PM, adriano wrote: Might just mean operators working for the company that connect with voip to the system and then take calls. (the old way of doing this was centrex in a hunt group and people taking calls at home) I think that the mobile operator as any other company receives calls by pots lines as T1 E1... ina ny way if he will receive calls through a gsm gateway the gateway itself must connect to pbx in a standard way probabilly voip the hardware will be a server an the interface .. hth Adriano Il 09/01/2013 18:38, C. Savinovich ha scritto: What in the world Asterisk to a mobile operator means? you mean you are are using a gsm gateway? what interface are you using?... not that I intent to answer your question, but you should be clear and specific if you expect someone to give you a pointer. Christian Savinovich */VoIP Telephony Consultant/* 646-982-3572 Original Message Subject: Re: [asterisk-users] IVR platform for a mobile operator From: Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com Date: Wed, January 09, 2013 10:07 am To: 'luke devon' luke_de...@yahoo.com mailto:luke_de...@yahoo.com, 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *luke devon *Sent:* Wednesday, January 09, 2013 9:06 AM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* [asterisk-users] IVR platform for a mobile operator Hi Friends , I want to setup a IVR platform using asterisk to a mobile operator. Can somebody give me some guides with recommended hardware types ? Thank you Luke. IMO you will be happiest with a SIP trunk handling this as there can be horrible latency in DAHDI/Mobile connections. -- _ -- 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 byhttp://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
Re: [asterisk-users] Paging unit suggestions
On 01/07/2013 03:41 PM, Doug Lytle wrote: The blowing fuses could be related to spikes etc., from a poor connection to the source, or a problem with the source hardware. If the amps are good, you could just drive them from a cheap phone with a regular headset jack They aren't, seem to be blowing fuses more often. Thanks! 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] Top Posting
On 01/02/2013 12:20 PM, Steve Totaro wrote: good one - me too ! On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro -- _ -- 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] Top Posting
On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Are you sure about that ? I know I was doing stuff with asterisk back in the LSS days and that was around 2001 Regards, Patrick -- _ -- 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] Top Posting
On 01/02/2013 03:35 PM, Jim Lucas wrote: On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! maildroid has a lot of features but kills your battery FAST. I only start it when I am expecting an important email and task kill it afterwards or it stays running. -- _ -- 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] Top Posting
On 12/30/2012 03:54 PM, Benny Amorsen wrote: Boy what an elitist attitude. I have been on this list far longer than most people - long before digium even existed and if you don't value what I have to say - well just don't read it. If you or your mail reader can't slice and dice a mailing list the way you want to see it well maybe its your opinions us top posters won't miss, since clearly you are lacking the skills to even have your tools format documents for you. Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. It is often a quick way to see if it is worth responding to someone. If they top post, nothing of value is likely to come out of the conversation. So by all means, everybody who wants to, keep top posting. /Benny -- _ -- 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] Paging for Praying
On 12/28/2012 08:13 PM, Steve Edwards wrote: Please stop saying don't top post, some of us prefer it that way. Please don't top-post. If you don't know what that means, please consult Google. On Fri, 28 Dec 2012, bilal ghayyad wrote: I have one more question: What was u meaning by call file and why it is required to place them in the 'astspooldir.'? There are 2 methods of originating a call external to Asterisk: call files and the Asterisk Manager Interface (AMI). A call file is a text file that you create. The format is very specific. It could contain (in the context of your needs) the phone number to dial and the path of the file to play. A call file is kind of like a 'message in a bottle.' You cast it into the sea and hope for the best. When this file is mv'ed into the directory specified in the Asterisk astspooldir variable, Asterisk will read it and try to do what you want. You can 'schedule' a call file to be processed in the future by setting the file's 'mtime.' The Asterisk Manager Interface (AMI) is a TCP connection between your program and Asterisk. You can issue commands (like originate) and receive responses. AMI is more robust because you can make decisions based on the response. If robustness is not of primary importance, a script scheduled by cron to run after midnight could create the 5 call files for that day, setting the 'mtime' of each file before mv'ing the file to the directory specified by astspooldir -- usually /var/spool/asterisk/outgoing/ How many customers will be receiving these reminders? What religion is this targeted to? -- _ -- 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] Hacked by Microsoft?
On 11/28/2012 11:52 PM, Steve Totaro wrote: You're not serious right ? That is just the center of the country since no better location is available. On Wed, Nov 28, 2012 at 7:45 PM, J Gao j...@veecall.com wrote: This morning someone tried to make sip call through my Asterisk. My server just drop these calls and record them in CDR with IP address: 2012-11-28 06:30:51 SIP/216... 10001000 1000 Hangup 999011972592249388 ANSWERED00:01 Hacker: 168.63.67.239 2. 2012-11-28 06:30:49 SIP/216... 10001000 1000 Hangup 88011972592249388 ANSWERED00:01 Hacker: 168.63.67.239 3. 2012-11-28 06:30:46 SIP/216... 10001000 1000 Answer 99011972592249388 ANSWERED00:02 4. 2012-11-28 06:30:43 SIP/216... 10001000 1000 Answer 1011972592249388 ANSWERED00:02 5. 2012-11-28 06:30:39 SIP/216... 10001000 1000 Hangup 2011972592249388 ANSWERED00:00 Hacker: 168.63.67.239 6. 2012-11-28 06:30:33 SIP/216... 10001000 1000 Hangup 7011972592249388 ANSWERED00:01 Hacker: 168.63.67.239 7. 2012-11-28 06:30:30 SIP/216... 10001000 1000 Answer 8011972592249388 ANSWERED00:03 8. 2012-11-28 06:30:27 SIP/216... 10001000 1000 Hangup 9011972592249388 ANSWERED00:06 Hacker: 168.63.67.239 9. 2012-11-28 06:30:25 SIP/216... 10001000 1000 Answer 011972592249388 ANSWERED00:07 Now I noticed something interesting: The hacker's IP address: 168.63.67.239 whois gave me: NetRange: 168.61.0.0 - 168.63.255.255 CIDR: 168.61.0.0/16, 168.62.0.0/15 OriginAS: NetName:MSFT-EP NetHandle: NET-168-61-0-0-1 Parent: NET-168-0-0-0-0 NetType:Direct Assignment RegDate:2011-06-22 Updated:2012-10-16 Ref:http://whois.arin.net/rest/net/NET-168-61-0-0-1 OrgName:Microsoft Corp OrgId: MSFT-Z Address:One Microsoft Way City: Redmond StateProv: WA PostalCode: 98052 Country:US RegDate:2011-06-22 Updated:2011-06-22 Ref:http://whois.arin.net/rest/org/MSFT-Z hmmm Did I just hacked by Micro$oft? Gao http://iplocation.truevue.org/168.63.67.239.html -- _ -- 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] Simple failover configuration
On 11/15/2012 10:27 AM, Eric Wieling wrote: What I have found most difficult in any failover situation is having everything decide at the same time something has failed. (this applies to anything not just asterisk) For example how does the polycom react if it can make the sip connection, but no outbound routes are available on the primary server for some reason ? Is your setup smart enough to actually shut down asterisk completely if its upstream network interface or route is dead to prevent local connections ? what if both servers are in that situation ? would they both shut down, neither ? what would you want to happen in that case ? They are not trivial questions to answer and the answers depend on your setup, there is no univeral right way of handling it. Polycom phones after firmware 2.x register to BOTH the primary and backup servers. On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? -- _ -- 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] Simple failover configuration
On 11/15/2012 10:31 AM, Danny Nicholas wrote: ran into this before on routers, you can put something like that or vrrp or carp in front of a pair of systems to fail to the right one BUT there isn't only one interface on something like a pbx, it has a lan interface and a wan interface, you have to get failover happening on the ALL at the same time which is tricky since when you fail one interface that sometimes eliminates being able to trigger the other interface to fail over since it no longer has any access to get a signal there. If you want to be able to look at the system to see why it failed you have to have some other way in as well. Every requirement just makes it more complex, and more fun to test. You can actually configure at least some Polycom phones to 3 or more SIP servers. Your problem is going to be that when one of your servers is down for whatever reason, the line key attached to that server will be off. In a Dual Server environment, I would lean toward putting something like Kamailio (sp) in line so it can determine which server is the active one. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, November 15, 2012 9:27 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple failover configuration Polycom phones after firmware 2.x register to BOTH the primary and backup servers. On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? -- _ -- 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
Re: [asterisk-users] USB FXS device
On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote: On 10/31/2012 01:44 PM, Russ Meyerriecks wrote: On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) Xorcom's Astribanks have native support in DAHDI http://www.xorcom.com/telephony-interfaces/astribank-usb-channel-banks.html Yes, but that goes against the spending a gazillion dollars requirement, and though I didn't specify my needs, I am just looking for a single FXS port. Basically I would like to build an ATA out of a Raspberry Pi :) Ideally for $100. why punish yourself like that ? pap2t is 2xFXS hanging off a network jack for $50 Half your budget, twice the density, and a nice box too Cheers, j -- _ -- 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] USB FXS device
On 10/31/2012 02:38 PM, Jeff LaCoursiere wrote: why not just get a usb headset and use with one of the sip client apps ? if you're going to the trouble of having a phone to plug in the fxs why rely on the pc at all ? use one of the spa type routers and plug the pc into it and the phone or if you have a free network jack just use a pap2t Then it works whether you have to reboot the pc etc and does not steal cpu cycles from the pc. Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) I found this: Broadtel UPA-1. I have email inquiries into them, but I saw in a blog post that they would provide Linux drivers on order, but nothing further... Cheers, j -- _ -- 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] high capacity analog - sip gateway
On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I'm looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- 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] high capacity analog - sip gateway
On 10/25/2012 05:09 PM, Carlos Alvarez wrote: On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank -- I'll look into that as an option as well. You should be able to also connect the Centrex lines to the channel banks, I believe. Best to check the specs of the actual phones, around here some of them are norstar phones that I am pretty sure are some sort of isdn (bri) thing rather than being a pure analog device. Better still take one of them and plug it in a raw analog line someplace and see what you get. I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] high capacity analog - sip gateway
On 10/25/2012 05:01 PM, Steve Totaro wrote: That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. I have had channel banks fail yes, and I stick by my assertion that failing a small $50 box is a lot less painful on the wallet and users than a channel bank with most of the extensions on it, this changes as the scale goes up though. I would only use a channel bank where the size can justify at least 3 of them, and I would never use a T1 based one again I would use the ethernet to FXS ones. I use a combination of analog and voip phones and there are various reasons for each being the type it is, one solution doesn't always fit everything, even within a single system. Get Adtrans, buy a four port T1 card or even better get the redfone device and do HA Linux between to boxes, you have immediate failover. http://www.red-fone.com/products-new/80.html I seriously doubt any product on the market is as solid, tried, and true as the traditional channel bank. You can pickup these channel banks very cheap used, and often find them in telco closets that have been abandoned. Thanks, Steve Totaro On Thu, Oct 25, 2012 at 4:29 PM, jon pounder j...@inline.net wrote: On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- 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] Fujitsu or Mitel PBX's
We are looking to find someone that is familiar with Fujitsu and Mitel PBX's. Email ru...@inline.net off list. 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] A new hack?
Well, that means opening up VPN connections from everywhere. Thats why I suggested turning off the server completely. hmmm - I thought that was the point of a vpn -- _ -- 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] A new hack?
On 11/30/2011 09:01 AM, Tom Browning wrote: I agree - its a bad comparison of 2 different things meant for different purposes. iptables is enforcement, fail2ban is detection. if you have time to sit and make up iptables rules by hand during every hack attempt 1) you have too much time on your hands 2) you have too much time on your hands On Tue, Nov 29, 2011 at 4:44 PM, john Millicanj...@millican.us wrote: Maybe I am misunderstanding the gist of the comment OP offered an invalid comparison of how iptables is better than Fail2Ban. Whether or not OP knew that Fail2Ban simply feeds rules to iptables is unclear from his comments. Log scraping is a time honored and effective method to correlate bad behavior. Log scraping can see things that no iptables rule would ever find. Think SSL. If Fail2Ban is a bad log scraper framework, then criticize it with a clear understanding of its role. -- _ -- 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] android won't play wav49: how to change format
On 11/25/2011 06:39 PM, Michelle Dupuis wrote: There is a script on www.generationd.com designed for Asterisk. It will convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc. and then email the message. It's a one line change to add to asterisk - very handy. (We use it for Android phones, nice to see call info in the tags as it plays!) what client app are you using ? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Friday, November 25, 2011 5:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format Wav49 is GSM wrapped in a MS header. You should be able reverse the order of the two items without harm. If you remove formats, then Asterisk won't find the existing messages or greetings in the format you removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 25, 2011 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] android won't play wav49: how to change format android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says Don't Change the Format Unless You REALLY Know What You're Doing! Well, I don't. Would this change screw things up? It's still the same formats, just a different order. sean -- _ -- 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
Re: [asterisk-users] Grandstream GXP2000 - copy configuration from handset
On 10/09/2011 09:52 AM, Silverthorne Wystead wrote: wget each of the screens I think there are only 6 Hello Folks; I may be posting this in the wrong list, but here goes. I have a Grandstream GXP2000 and I would like to use tftp or some other utility to grab the configuration from it. Anyone have any bright ideas? Thanks Glen -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 04:41 PM, Alex Balashov wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- _ -- 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] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 08:47 PM, Bruce B wrote: This is becoming just like the bacula mailing list where anyone that knows anything is beaten into submission for daring to question the great and powerful oz. You are very childish besides being very useless. Also, note that there are others that are bothered by the same changes that are uncalled for. I was as constructive as possible but you think starting a sentence with I am not trying to be rude... is rude. LOL. I have said that upfront so idiots like you don't take offence but you did and you read as, I am trying to be rude Well, suit yourself and keep sucking up Alex. On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 tel:%2B1-678-954-0670 Fax: +1-404-961-1892 tel:%2B1-404-961-1892 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 -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 02:02 PM, linux guy wrote: get any cheap android device and load linphone. or grandstream works for a wired device. gxp2000 has enough line buttons you can easily route calls for multiple people to a phone and tell who the call is for I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. Any ideas ? Thanks ! LG -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 02:18 PM, linux guy wrote: In our house, we need wireless. I have a Grandstream already. I am looking for something with a form factor more conventional than a cellphone. Maybe that is silly ? I see various unlocked large screen Android devices for ~$150. not sure what you mean more than a cellphone ? get an ata such as an SPA series with 2 channels on it or PAP2T and plug a cordless base station in each one if you want to go cheap. it works but a single line cordless does not at all do justice to what can be done with asterisk. When you say large screen are you talking about an APAD type thing? they work just fine mounted on the wall (I use them for intercom and general control system interface) but I would not walk around holding one to my ear. I was hoping to spend on the order of $50 per handset. I don't understand why (other than economies of scale) that I can buy various wireless POTS wireless systems (base and multiple handsets) for $100 and yet a single VOIP wireless handset is $150. Any comments on integrating a wireless POTS system into an asterisk system ? -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On 08/26/2011 02:26 PM, Jeff LaCoursiere wrote: On Fri, 2011-08-26 at 12:10 -0600, linux guy wrote: I was thinking of using a PAP2T-NA for the ATA to handle the fax. It appears to have a large number of fax specific settings. Can anyone comment on using this device with a fax ? If you are using POTs to bring in your fax calls you should be ok for home use. I do this with a PAP2T-NA, Hylafax, and iaxmodem. I have iaxmodem accept the fax, then relay it to the PAP2T. I use the second port to drive a Panasonic DECT base station with five satellites, which I have spread around the house. The Panasonic is the only one I have found that has the ability to host a ton of satellites without having a ton of redundant features on the base station (don't really need an answering machine, for example). Its also the only one where the handsets have nice 2.5mm headset jacks, which I use all day. I've never really understood the need for wireless SIP handsets. An ATA plus a normal DECT set seems perfect to me. The way I use it I have one device in my pocket, I can get my calls on the couch, in the yard, down the street, at the office or in another city. sip + wifi doesn't just have to be at your house, it can be anywhere there is wifi available. j -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 03:17 PM, linux guy wrote: I like the idea of running multiple ATAs with a single base or handset on each line. Something like the Panasonic KX-TG4111B which sells for about $40 for a handset and base. PAP2s sell for about $50 or $25 per line. Total cost of $65 per handset. Comments on this approach ? I use it mixed in with other types of devices. The pap's really are pretty flexible devices for the money. I have one cordless but mostly wired fax, and doorphone. I have a rack of them and each just terminates into the building wiring for phone jacks, so at the user end its just a regular phone plugged in the wall. I used to have t1 card and channel bank, but that became a single expensive point of failure, needs a pc that has pci slots, and really is no more programmable than the sip boxes. So that is why I switched. The cordless handsets I just don't really like for reasons I have gone into already but that doesn't mean they don't work. -- _ -- 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] Looking for ideas for nice home phone system
On 08/22/2011 04:11 PM, Linuxguy123 wrote: I have a home and business system that just ties all the lines together (combo of zaptel, and sip incoming at several locations), inbound routing based on which line it came from. Was using a t1 card and channel bank for extensions but migrated away from that to a mix of multiline sip hardphones, sip softphones (touchscreens, as well as clients on pcs and wifi on android phones), as well as good old analog phones hooked up to SPA's (they are all in 2 central locations and just use legacy phone wiring), got physical fax machines hooked to those as well for outbound, inbound fax on all lines to iax softphone and hylafax, even door phones on the spa's in immediate answer mode. Second dial tone provider for ld on the physical lines all hidden by the way dialing is done. Conference rooms, speed dials, voicemail unified messaging. Even have forwarding like : - incoming call rings local phones and sip on wifi if its connected (to android phone) - call rolls over to call out another trunk and dial cellphone (hiding this from inbound caller entirely) - still no answer rolls back in to unified messaging voicemail so no messages are ever left on cell phone and subscribing to vm not even necessary. From my android handset any call I dial I can choose to use cellular or sip to complete the call. Works just fine, really have no major complaints about it. Use the freepbx gui. I'm looking for ideas for building a innovative, powerful home phone system. Something that does voicemail well, integrates cell phones into the house system, etc. I know there are a lot of details that need to be discussed, but lets leave it at that for now. What is everyone doing ? 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 -- _ -- 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
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. It was because you were setting up the canopy stuff, not related to being in Iraq. Motorola sure goofed on that abortion. 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 -- _ -- 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] Free CNAM
On 05/29/2011 09:37 AM, Michael R. Wally wrote: So how long till its an adaptive telemarketing blocker based on the query velocity of the numbers ? The system uses real Telco CNAM Dips. Any generic names you get are from the subscriber's carrier itself. We can only provide what we ourselves get. I tried it, but it returns the same kind of junk that some of the databases do. For example, on a Florida number, it just says FLORIDA instead of the proper name (some of the CNAM databases have the right name). -- _ -- 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] MagicJack quality
On 05/24/2011 11:35 AM, A J Stiles wrote: Someone asked about the quality of it, he was quoting the hardware specs of a similar device. I doubt magicjack publishes that kind of detail about theirs. so where is the problem ? Its irrelevant he represents that device commercially. On Tuesday 24 May 2011, BroadTel wrote: Hi all, Just in case if anyone will be interested in *REDACTED*, a USB to FXS adapter embedded with SIP softphone. Product specification is as follows: Please refer to the fifth and sixth words of the title of this mailing list. To everyone else, I recommend $ echo -e :0\n* ^From.*BroadTel\n/dev/null .procmailrc -- _ -- 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] MagicJack quality
On 05/24/2011 11:57 AM, A J Stiles wrote: On Tuesday 24 May 2011, jon pounder wrote: On 05/24/2011 11:35 AM, A J Stiles wrote: Someone asked about the quality of it, he was quoting the hardware specs of a similar device. No they didn't. The original message to which the spammer was pretending to reply (and in the wrong place, even) never existed in the first place. I doubt magicjack publishes that kind of detail about theirs. so where is the problem ? Its irrelevant he represents that device commercially. The problem is that it is a blatant advertisement, not-very-cunningly disguised as a legitimate response to a question. This is known as tag-team spamming (or at least, it is when someone actually pretends to post a question). At least its a related product, not porn or pills. -- _ -- 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] MagicJack quality
On 05/24/2011 02:45 PM, Warren Selby wrote: On Tue, May 24, 2011 at 12:33 PM, Doug Lytle supp...@drdos.info mailto:supp...@drdos.info wrote: Steve Edwards wrote: My archives don't go back that far Mine do. No match on Jack, magic or itntelecom.com http://itntelecom.com mailto:c.savinov...@itntelecom.com mailto:c.savinov...@itntelecom.com Doug http://www.mail-archive.com/asterisk-users@lists.digium.com/msg207540.html A quick google for magicjack quality C. Savinovich turned this up as the second hit...that being said, I agree the OP from today is just spam and doesn't really help anyone. -- Thanks, --Warren Selby, dCAP When you get right down to it, the following is just as spammy and it doesn't even describe what is at the link. I don't agree with what the other guy did, but if you quickly scanned the email you knew all the details without having to go to some other site. If I want to know what selbytech is about I have to click there, and this link is injected in the mailing list over and over. So don't be so quick to criticize someone when you are doing things just as bad IMO. Our website just got a facelift! Check it out! http://www.SelbyTech.com 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 -- _ -- 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] Android tablet voip?
On 04/01/2011 11:00 AM, Dean Collins wrote: Anyone on the list using and Android tablet with a voip service as their primary phone device? Either with wired or Bluetooth headset? What are you using hardware/software. What are your thoughts? I use linphone. Config : - incoming ivr etc to get to extension - incoming call rings sip extensions for me including android if it happens to be registered. - if no answer call goes back out pstn to cell number (no voicemail on cell) - if no answer on cell, asterisk gives up and goes to its own voicemail issues : - linphone is the only sip client I found that actually reliably worked and was able to register, pass audio, dial and so on. (its not the most glitzy client though) - linphone can be set to only register on wifi - this is fine but it stupidly still advertises its intent to handle calls when its not registered, so when you dial generically you get a popup asking whether to use dialer or linphone to place the call (that's just bad programming). - due to the way sip works and the speed things happen, the phone can still be ringing the sip call when the cell call comes in and there is a moment when its hard to answer the right thing since both are ringing but the first caller is already gone. overall works adequately, I am able to use it like a sip cordless phone when around wifi, and transparently as a cell when not. the fact its a 2 stage ring to the same device when in wifi range makes it more confusing than if both had separate numbers completely. I also have an apad wall mounted running linphone among other stuff, and it functions well as my doorphone - linphone rings and pops to the foreground when a door call comes in. its also running on wifi but its sort of a fixed application, not what I would really consider a phone. Cheers, Dean -- _ -- 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] Testing from where number is...
On 03/03/2011 03:53 PM, Danny Nicholas wrote: Not having an in-depth knowledge of how EU numbering works, I would still suggest that you could get pretty far with the numberingplans AGI if you made a database that blocked out the number once it came up as a cell. In the U.S. cell phones have their own local code, IE 205616 is going to be a cell so you can eliminate 205-616- thru 205-616-. with number portability in north america, even knowing the npa npx doesn't really help you, you can port a landline to a cell or vice versa, you can tell what carrier/media the number was originally allocated from but if it was ever ported that information is meaningless. Granted not a lot of people port numbers since you still have to argue to get it done around here (in Canada) but its been the law for several years if its still a local exchange the porting costs $30 and they have to do it if you ask. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Piotr Górski *Sent:* Thursday, March 03, 2011 2:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Testing from where number is... As free I mean no subscription. I can write AGI that will query numberingplans.com http://numberingplans.com - that's not a problem... but I can query site only 20 times a day without a subscription... So it's not free. Z poważaniem, Piotr Górski, CONCEPT MUSIC ART SP. Z O.O. ul.Dauna 70 30-629 Kraków http://www.cma.pl pi...@prnet.pl mailto:pi...@prnet.pl GSM: +48 609 127 370 Faks: +48 12 444 1051 2011/3/3 Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com To beat the dead horse, if you want it free, write an AGI... *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Piotr Górski *Sent:* Thursday, March 03, 2011 1:58 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Testing from where number is... Something free? -- Piotrek Gorski 2011/3/3 Faisal Hanif fai...@vopium.com mailto:fai...@vopium.com www.numberingplans.com http://www.numberingplans.com -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski Sent: Thursday, March 03, 2011 12:02 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Testing from where number is... Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: Cell Phone, Belgium or Land Line, Germany. -- Piotr Gorski -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Recommended Windows client to display CID?
On 01/26/2011 08:52 AM, Gilles wrote: If you like open source what are you doing running windows ? Getting anything to work properly there which does network communications is a huge PITA since every user has their own firewall and different settings etc etc etc. Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. -- _ -- 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] Mailing list question
On 01/20/2011 12:01 PM, Andrew Thomas wrote: why not just subscribe with an account that doesn't do that like gmail or yahoo ? Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something likedisclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Top Posting
Surely there is some mail client smart enough to be able to flip around the levels of indenting so most recent is top or bottom. If not quit bitching and make one - I will continue top posting since I don't seem to be alone in preferring it. On 01/16/2011 10:28 PM, Mark Murawski wrote: We obviously have all our own opinions about being on top or bottom. And it boils down to personal preference obviously. I think in all cases, top posting is by far superior. But I think the battle will continue ad infinitum. One, because of speedups in finding the most recent content which will always be on the top. Two, with the new content on top, reading list posts from any phone becomes really easy. If you have a particular thread you're following, you can quickly look at the new reply without having to do anything other than open the email! I don't know of any phone that's 'smart' enough to auto scroll to the bottom when you open up a list post where someone has bottom posted. On 01/16/2011 10:17 PM, Tilghman Lesher wrote: On Sunday 16 January 2011 20:47:56 James Miller wrote: When you get over 500 emails a day on your blackberry you have make a decision on what is or is not worth reading at that moment. Clearly, then, the problem is your blackberry. Ditch it. Or stop subscribing to list email on a device which is clearly not up to the task. Or would you say that since it's inconvenient for you to clean up your dog poo, you shouldn't have to pick it up? And leave it where the rest of us might step in it? If you cannot be bothered to clean up after your dog, maybe you shouldn't be taking your dog to the park. Similarly, we may not be able to fine you for failing to obey list rules, but the rules still apply to you, like it or 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 -- _ -- 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] alarm POTS lines
On 12/04/2010 03:01 AM, Tammy A Wisdom wrote: You all do know that this is something you could be sued for since this is 'life safety equipment' ? I've heard from multiple sources that if it isn't against the nfpa that it will be very soon Ask yourself if you want to be subject to a lawsuit if someones house is robbed, someone dies, house/business burns down and the alarm panel is unable to communicate? Ask your insurance rep if they will cover you doing this type of voip stuff and they'll most likely tell you not too. Bottom line is this is NOT a smart thing to put on voip. Tammy Tammy A Wisdom Summit Open Source Development Group nfpa only applies where nfpa is actually a requirement, ANYTHING anyone does that is not required (residential fire and burglar alarm) is better than the alternative of nothing at all. Calling yourself a ulc certified monitoring center or something like that is a no no if you don't meet those requirements, but selling a service as what it is, is perfectly fine, if you are not guaranteeing something you can't provide you are in the clear. All that aside things are moving to direct tcp/ip communication, using voip to talk to antiquated monitoring equipment is just an interim fix as the consumer end technology is moving faster than the central station technology. The argument voip is unreliable compared to pots really is not true if everything is setup properly. tcp/ip and anything that lives on top of that protocol like voip etc., has much more flexible routing and failover than any pots circuit could ever hope to have, and can use multiple independant paths for communication, and the connection can be held open like a dvacs circuit for continuous monitoring without all the overhead of individual copper pairs and multiplexers at the monitoring center. Its hard for the manufacturers of this equipment to stomach though that a pc with a network card is actually more functional than a multimillion dollar piece of hardware they used to sell, so change is slow coming. -Original Message- From: Ryan Wagonerrswago...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 4 Dec 2010 00:03:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] alarm POTS lines On Thu, Dec 2, 2010 at 11:58 AM, Jeff LaCoursierej...@sunfone.com wrote: Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a fax relay setup locally with hylafax/iaxmodem to solve that issue, and it is working very well. We don't however, have a solution for their alarm lines. The problem is of course that modem calls over VoIP are flaky at best. Even though these alarm calls are low baud rate, when we test with the alarm company we only pass about 30% of the time (ulaw from customer site to our central switch, then out a T1). To be fair there is no QoS on their Internet links yet, and that certainly plays a role. But it seems to me that there should be a solution much like our fax relay, where we literally accept the fax call over the local LAN, produce a PDF file, transfer it to the central switch which then dials it back out over a T1. In that case the only modem over VoIP is on their local LAN, which has performed well for us. I would love to see a DSP modem that could answer an asterisk channel, send the data stream over TCP to some remote asterisk, which could then relay the stream by making an outbound DSP modem call on a PSTN trunk. Has anyone attempted anything like this? As an aside, since the recent thread on Seagate Dockstar installs, I have several running. This would be the perfect platform for the relay on the customer end, being so ridiculously cheap (I bought three for $30 each, plus 3 $10 4G USB sticks). So hoping this will spark some comments on the concept in general, and really hoping someone has actually tackled something similar. It could open up a nice niche for even residential customers with expensive POTS lines dedicated to alarm systems. Cheers, -- Jeff LaCoursiere SunFone j...@sunfone.com Alarm panel communication, at least with Ademco, is done only with DTMF. When I tested the Asterisk AlarmReceiver application I found that the DTMF tones were so short they weren't always recognized by my ATA in RFC 2833 mode. Changing it to inband DTMF worked better, but then I was having issues with the AlarmReceiver application. Have a look at the below link, which touches on alarm panel communicates.
Re: [asterisk-users] Someone has hacked into our system
On 11/22/2010 06:44 PM, Kevin Keane wrote: Use IPTables to lock down your machine to only accept incoming connections from your local network and from the particular IPs that you are expecting connections from (such as your SIP trunk, maybe). That is of course assuming that these calls are made by SIP. Don't forget to also change all the passwords. good point - someone can easily just dial in a pots line locally and dial out another one making a long distance call, assuming the dial plan allows this. it doesn't have to be sip involved in any part of the problem. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Kuznitz *Sent:* Monday, November 22, 2010 8:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- 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] Recommended *WRT router to run Asterisk?
On 11/19/2010 10:10 AM, Michael Graves wrote: On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote: Interestingly for commercial units, I've had the opposite experience - I've found that my (business) customers just will not pay for something tiny that's capable of supporting 30 phones... I did have a look at the GuruPlug stuff recently, but it's just not going to be sensible for me to put any effort into it as people won't buy it. Even my smallest box at 150mm square is verging on the unbelievable for some people - especially those used to a PBX taking up a whole rack, or having 2 or more large units bolted to a wall... Still, for home/hobby use these little things are great! It seems to me that there's definitely a break point below which very small hardware platforms are really only suitable for hobby or very niche applications. I once ran Astlinux on a Gumstix board just to see if it was possible. But beyond embedded applications it actually creates problems to have such small hardware. OTOH, net-tops like the the Fit-PC2i are really very interesting, and servicable in small producti roles. http://www.mgraves.org/2010/07/d-i-y-asterisk-appliances-a-question-of-s cale/ The problem is people want to find the $50 embedded solution and then use it where the $1000 solution is really needed. What is nice is when the $50 hardware and the $1000 hardware run exactly the same software so other than the drivers for the hardware itself, everything else behaves the same way and its easy to move around configurations to grow. (I am not talking about asterisk specifically, just generally about routers, backup devices, media servers, etc) Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] Recommended *WRT router to run Asterisk?
On 11/18/2010 10:02 AM, Chris Gentle wrote: On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr mailto:codecompl...@free.fr wrote: Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: I'm running Asterisk 1.4 on a WRT54GS that I picked up off ebay for $50. The WRT54GL doesn't have quite enough memory so I went with the GS model. I'm running OpenWRT on it. I was mostly experimenting with it but ended up installing it at my parents' house as a kind of batphone solution. I also hung a couple of SIP phones off of it giving them a couple of different extensions, one of which works across a WIFI connection. Their WRT54GS connects to my Asterisk 1.8.0 machine using IAX. Both endpoints are behind NAT. Works pretty well for me. -- Chris I have a similar setup in an office but sip directly back to the main server - not sure what the value add to the local asterisk is, except intercom calls would not have to leave the lan, but isn't that the purpose of reinvite ? -- _ -- 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] Door Contacts via Asterisk?
On 11/15/2010 02:49 PM, Mark Scholten wrote: Anyone have a soft sip endpoint which can take touchtones over sip and run scripts ? that is a more general purpose integration solution to asterisk itself. I realize there are scripts for dialplans which can do this already but often the door is nowhere near the core asterisk server. Hello, We did something like that in the past (but for 1 company, but it shouldn't be really different). The easiest solution for us was to use a door opener that could work with almost any normall phone connection and use a Linksys pap2t or something similar. With kind regards, Mark Scholten *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Cassius Smith *Sent:* Monday, November 15, 2010 7:35 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Door Contacts via Asterisk? Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- 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] FW: Under heavy attack
I'm still on old-fashion copper-wire and have yet to experience the joy of SIP Trunk-ing and the type of issues discussed in this thread. My thought to share here is that outgoing calls should be easy for thoroughly authenticated users and impossible for others... Probably more can-o-worms than help. Sorry if this is so. nothing new here, this is just the digital equivalent of a wats line with a weak access code for outbound access. the difference is code guessing can be a lot more aggressive now, and finding the inbound path is simpler. -- _ -- 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] FW: Under heavy attack
On 11/01/2010 01:44 PM, Nyamul Hassan wrote: I think the only real solution here is to make people take more responsibility for their actions - find and punish the actual abusers - make users liable for damages caused by infected PC's - defaults from an isp should be everything locked down but with user able to request more ports being opened at no extra cost, if a user asks for it they then take on responsibility for the use of that port. LOL On Mon, Nov 1, 2010 at 23:33, Cary Fitch ca...@usawide.net mailto:ca...@usawide.net wrote: I was going to point out a failing of the attackers, but figured they read the list and don’t need any more tips. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Monday, November 01, 2010 12:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FW: Under heavy attack And obviously these attackers read our emails on lists like this and adjust their sick strategies accordingly. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com www.pbxforall.com http://www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com mailto:jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net mailto:jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- 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
Re: [asterisk-users] FW: Under heavy attack
On 11/01/2010 08:20 PM, Joel Maslak wrote: Be careful, telcos may make the users responsible if they have insecure PBXes...right now they often write off much of the charges. you must have a great telco - around here even credits that are agreed to seldom show up on the bill without investing more time chasing it than the credit is worth. But I do agree that there would be a lot less garbage on the net if everyone was liable for their insecurity. Heck, there would be no SIP attacks if everyone's systems were secure - there would be no gain in trying to exploit reasonably unexploitable systems. On Nov 1, 2010, at 11:54 AM, jon pounder j...@inline.net mailto:j...@inline.net wrote: On 11/01/2010 01:44 PM, Nyamul Hassan wrote: I think the only real solution here is to make people take more responsibility for their actions - find and punish the actual abusers - make users liable for damages caused by infected PC's - defaults from an isp should be everything locked down but with user able to request more ports being opened at no extra cost, if a user asks for it they then take on responsibility for the use of that port. LOL On Mon, Nov 1, 2010 at 23:33, Cary Fitch ca...@usawide.net mailto:ca...@usawide.net wrote: I was going to point out a failing of the attackers, but figured they read the list and don’t need any more tips. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Monday, November 01, 2010 12:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FW: Under heavy attack And obviously these attackers read our emails on lists like this and adjust their sick strategies accordingly. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com www.pbxforall.com http://www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com mailto:jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net mailto:jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- 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
Re: [asterisk-users] Under heavy attack
I already have a monitor (tied into nagios, which pages me if my fraud thresholds are exceeded), but I feel that is probably beyond the abilities of most of the people experiencing call fraud. The people who know what they are doing with Unix and Asterisk are generally not the victims of this. It would be nice if there was something built into Asterisk to alert on fraud - something that an end user with little Asterisk (or Unix) experience could utilize to be alerted to call fraud, which is easily detectable almost 100% of the time (too many calls for the organization == call fraud). And that is really what this is about - keeping someone from getting a $30,000 phone bill. It certainly should be the part of any commercial offering. what are you using that is tied to nagios ? -- _ -- 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] Under heavy attack
On 10/31/2010 11:39 AM, Mark Deneen wrote: On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslakjmas...@antelope.net wrote: If these are mobile users, I hope they never use any public networks (hotels, starbucks) where other subscribers can do things like ARP attacks to do MITM (and steal your calls; it might not be happening today, but it will be happening soon - as the social networking attacks demonstrate). If you do have truly roaming users, I hope you use HTTPS (with validation of certs turned on) or a VPN (likely not an option of connecting to an ADSL site, due to bandwidth concerns). Can you explain why VPN is not an option for ADSL? (Open)VPN overhead is not that high. ~70 bytes per packet if I remember correctly. -M We're not using it for calls but do have a huge openvpn infrastructure connecting wifi access controllers and there is not a ton of overhead at all, and it runs on endpoints with very limited resources. What might need lots of tweaking is how the sip packets get converted to vpn packets and transmitted, since there could be a lot of fragmenting and reassembly. If phones came with it built in, the manufacturer would presumably have figured this all out for them. PPTP is another option thats widely supported but I don't have much personal experience with it. -- _ -- 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] Under heavy attack
On 10/31/2010 12:58 PM, Joel Maslak wrote: On Oct 31, 2010, at 9:40 AM, jon pounderj...@inline.net wrote: what are you using that is tied to nagios ? I'll package it up next week and make it available. Basically, I use nrpe to call a shell script that looks at the last five minutes, 60 minutes, and 1440 minutes of a asterisk -rx 'core show channels' output that I run from cron every minute (I count the number of paid channels in use [I ignore channels that have no cost associated with them, such as users calling other users]). If any of these thresholds exceeds my error threshold, I signal a nagios CRITICAL alert. Otherwise I return OK. ok thanks. btw - on the subject of nrpe - anyone got a version that runs stable on windows ? we have one but it randomly locks up (without failing the service) every week or so on various windows servers, service down detection in windows sees it up, and nagios can't use nrpe to restart it with a command since that is what is down. annoying when its 3am. on the linux boxes, works perfectly. -- _ -- 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] Under heavy attack
On 10/30/2010 04:07 PM, Stuart Sheldon wrote: any registry of abusers like for spam ? any list of complete ip ranges for countries where abuse is rampant to block ? I am getting sick of the one offs and ready to start blocking big chunks of address space. -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 We are also seeing an increase in attacks. And yes, there is a benefit to blocking them. They tend to go away if you have them restricted, where if you let them go at it, they will sit on your host for sometimes hours. Stu On 10/30/2010 12:43 PM, Joel Maslak wrote: Is there really any benefit to blocking these, if you use good passwords? On Sat, Oct 30, 2010 at 1:20 PM, Warren Selbywcse...@selbytech.com mailto:wcse...@selbytech.com wrote: I'm experiencing this on one of my clients servers. The attack is ongoing. Thanks, --Warren Selby On Oct 30, 2010, at 2:28 PM, Zeeshan Zakariazisha...@gmail.com mailto:zisha...@gmail.com wrote: My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- http://www.ilovetovoip.comwww.ilovetovoip.com http://www.ilovetovoip.com http://www.pbxforall.comwww.pbxforall.com http://www.pbxforall.com (beta) -- _ -- 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/hellohttp://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) iQIcBAEBCAAGBQJMzHsdAAoJEFKVLITDJSGS2fwP/j7/Jkcza71zoEMPMdegh+K5 ASVOda6yPazRmY6LAjqrNTwMyASmmngr/LLZbBmqRNXdzjWqDJ5+CEmCK09/WlcB etoz09XTNd0mswMq8r2uVSdKE7PBTZRlNokIfwbwSvWFIL01qbdA3urHVIJuNDuI V2eN94K+lgX7m69TFHe4J209X7BXQS3HxDl0aQVcW+NnofWj9o6BXoLdQXrkS/sG C7npBqpUe1asoyl2Bo5qSpzzMGiebZOcMIjKAEEu0anESZKKuNIhcj4BX6uOCRk0 8//IlNmqMVKfJr8ttpqZVbbKI9AKjTWBHV77LzSNkPgcFjD6WeiOSnOMWW0UNAgE 3iaTCzXO9GwJLhRucdoezCI78qCkFdO8N0C6UZcrW/eP7bJdxa4Ab0of3EtG3V2U QjeKQYYpL7O0my3uwO4I1BY7qiDTqibTzQ6Gb7Y4No029R78cWff3xIueU5rNZeO Fr/2ODNFZE0Q1+KA7d29308jIKY0Ubz5s/QBKbAjWfQk80dQ4BE/6nqBUJmZWIAx CNL8dK+jv6uCIi5Ae2tMHGestkcy4Ol4fdKC6emVLgm4DbRYKAg259lkoAifT7qo 8/0LWfjuP8mXHaQ2x023wTKg+FyZCIwJmpr8UDaKwMdtFgwpLuZeQrYuRQiW8TCS xkBSL1xkLIoEy1b3NLDv =eqQ+ -END PGP SIGNATURE- -- _ -- 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] Mobile Phones and Asterisk
On 10/30/2010 09:24 PM, Sebastian wrote: On 10/29/2010 04:40 AM, jon pounder wrote: On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote: Here is what I do today and it works fine: - asterisk/trixbox - Dext/android phone - Bell Canada cell provider - call comes in, to an extension with voicemail - rings a bunch of sip devices (real phones, and the android via linphone if it happens to be near wifi and registered (set to only use wifi not 3g to register) - if not answered call is forwarded back out a pots line and dials the cell number (cell is not subscribed to provider voicemail) This is an advantage over my situation. Here (UK) - if you don't configure voicemail on your mobile - the mobile operator just plays a message along the lines The phone number is not available right now. Please try again later (or something similar). Which screws things up - as Asterisk can't tell that the mobile is not available. To Asterisk, that message is the same as somebody answering the line. Same in France and Spain - as far as I've seen. I think it does that here as well, but after a much longer delay than asterisk sits around waiting - like close to a minute I think. It definitely varies by carrier as well - Rogers here can't even get their heads around delivering a txt message from an email to sms gateway, let alone handle something like the above. Sebastian - still no answer that pots line is hung up and call drops back into the original extension's vm. (I have not run into a problem with answer detection, only that people don't stay on the line long enough for me to answer on the second set of ringing, but if they are that impatient the call was probably not important anyway) outgoing calls if registered I have a choice once I dial of linphone or dialer to make the call. checking vm is just *98ext from linphone as the dialing app, or dial in and navigate to vm. linphone is a little less polished gui but seems to work the best for me to reliably register when it should. (tried about 5 different sip clients) Hi, Thanks for your very informative response. This is really helpful. I wouldn't be pushing it though since it isn't possible as of now. Kudos! RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Friday, October 29, 2010 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Hi, On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote: Hi, I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix. What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not. Maybe others who know better will jump in - but I seriously doubt you will be able to do this. From my limited knowledge, I believe mobile phone networks use different signalling then regular terrestrial based providers. I don't really think that the engaged tone sent back by the mobile operator will be decoded correctly by Asterisk. Not to mention that, I don't what happens where you are - but in UK for example - you don't even get an engaged tone from a mobile phone. You just get either sent to the user's voice mail, or you are played a message from the mobile phone operator which essentially tells you that the user is engaged or unavailable. Operators in many other European countries do the same. So from the point of what you are trying to achieve - this is useless in Asterisk. I would have liked to do the same thing - as I have line divert in Asterisk to my mobile phone - and I would have liked for Asterisk to just skip along to my Asterisk voice mail when my mobile is either out of coverage, or when I'm in a conversation on it. But no such luck. I believe the mobile operators wouldn't like the idea anyway - as they get to charge you extra for playing all those messages or sending you to their voicemail. I believe in parts of the North American continent things are similar, but even worse. As the caller gets charged as soon as the mobile phone starts ringing - apparently simply the act of accessing the mobile operator's network is chargeable - never mind if you get to speak to anybody or not. Then again, maybe things are different where you are - and maybe there is a way to get Asterisk to recognise the busy tone from your mobile operator. Maybe somebody here will jump in with a suggestion. It seems that it has to do with busy signalling in Asterisk. A softphone I believe will accomplish this out of band - with some commands over SIP. While PSTN (normal phone lines) and mobiles I believe tend to signal this with inband tones (part of the sound coming down the line). You might also want to check
Re: [asterisk-users] Under heavy attack
On 10/30/2010 11:25 PM, Warren Selby wrote: To me it seems the real question is What is going on today?. I normally get eight to ten asterisk-related fail2ban alerts a day between a few client sites - today I've received at least 10 times that many attacks on just one site. These are all coming in from different ip addresses, a new one every few minutes. These addresses are located all across the globe. This seems like some kind of coordinated assault - maybe someone is activating a 'bot-net' for sip attacks? Certainly looks like it to me, I am seeing the same thing. Thanks, --Warren Selby On Oct 30, 2010, at 9:02 PM, Andrew Lathamlath...@gmail.com wrote: They have agreements for termination to locations with high rates. These types of attacks happen on servers that fit a digital signature. With certain ports or certain versions of software on those ports. Yes the Art of War is required reading for todays systems administration professionals... Change your signature, change your ports. What are they after, anyway? Merely cheap international calls? -- Tzafrir Cohen -- _ -- 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] Mobile Phones and Asterisk
On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote: Here is what I do today and it works fine: - asterisk/trixbox - Dext/android phone - Bell Canada cell provider - call comes in, to an extension with voicemail - rings a bunch of sip devices (real phones, and the android via linphone if it happens to be near wifi and registered (set to only use wifi not 3g to register) - if not answered call is forwarded back out a pots line and dials the cell number (cell is not subscribed to provider voicemail) - still no answer that pots line is hung up and call drops back into the original extension's vm. (I have not run into a problem with answer detection, only that people don't stay on the line long enough for me to answer on the second set of ringing, but if they are that impatient the call was probably not important anyway) outgoing calls if registered I have a choice once I dial of linphone or dialer to make the call. checking vm is just *98ext from linphone as the dialing app, or dial in and navigate to vm. linphone is a little less polished gui but seems to work the best for me to reliably register when it should. (tried about 5 different sip clients) Hi, Thanks for your very informative response. This is really helpful. I wouldn't be pushing it though since it isn't possible as of now. Kudos! RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Friday, October 29, 2010 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Hi, On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote: Hi, I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix. What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not. Maybe others who know better will jump in - but I seriously doubt you will be able to do this. From my limited knowledge, I believe mobile phone networks use different signalling then regular terrestrial based providers. I don't really think that the engaged tone sent back by the mobile operator will be decoded correctly by Asterisk. Not to mention that, I don't what happens where you are - but in UK for example - you don't even get an engaged tone from a mobile phone. You just get either sent to the user's voice mail, or you are played a message from the mobile phone operator which essentially tells you that the user is engaged or unavailable. Operators in many other European countries do the same. So from the point of what you are trying to achieve - this is useless in Asterisk. I would have liked to do the same thing - as I have line divert in Asterisk to my mobile phone - and I would have liked for Asterisk to just skip along to my Asterisk voice mail when my mobile is either out of coverage, or when I'm in a conversation on it. But no such luck. I believe the mobile operators wouldn't like the idea anyway - as they get to charge you extra for playing all those messages or sending you to their voicemail. I believe in parts of the North American continent things are similar, but even worse. As the caller gets charged as soon as the mobile phone starts ringing - apparently simply the act of accessing the mobile operator's network is chargeable - never mind if you get to speak to anybody or not. Then again, maybe things are different where you are - and maybe there is a way to get Asterisk to recognise the busy tone from your mobile operator. Maybe somebody here will jump in with a suggestion. It seems that it has to do with busy signalling in Asterisk. A softphone I believe will accomplish this out of band - with some commands over SIP. While PSTN (normal phone lines) and mobiles I believe tend to signal this with inband tones (part of the sound coming down the line). You might also want to check your regional settings in Asterisk. Sebastian I achieved this successfully by emulating it via a softphone, when I call a softphone and it is currently engaged in a call, asterisk returns BUSY in DIALSTATUS and will automatically fallback to the next step in the dialplan. But this is not the case when applying it to the mobile phone. When the target phone is currently engaged in a call, and I called the mobile phone, I can hear a busy tone(which is alright, since the target phone is actually busy), but it will wait until it timed out as defined in the DIAL cmd, and the var DIALSTATUS returns NOANSWER, instead of BUSY, as if the mobile phone is available and it was not answered at all. It may also have to do on how the tones are being handled, or it can also be that the mobile phone and the media gateway are the one talking to each other, and asterisk cannot get the status of the phone itself.
Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
On 10/02/2010 03:31 PM, bruce bruce wrote: Hi, Can you please explain the DynDNS part. How would I put that in my Asterisk server as an identified party? Usually it comes to me with IP address (dynamic). Or do add something like this in sip_nat.conf: externip=mybox.dyndns.org http://mybox.dyndns.org localnet=192.168.0.0/255.255.255.0 http://192.168.0.0/255.255.255.0 every time the address changes you have to have some script to make the change in your firewall. ??? Thansk again, On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net mailto:j...@inline.net wrote: On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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] Purpose of qualify=yes
On 09/16/2010 12:01 PM, Chris Owen wrote: well that just means you need a trunked satellite pbx where all the phones are, and that would take load off the main connection. half those people have got to just be talking to each other and don't need to use the gateway at all. On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote: I prefer to keep qualify=on for all the extensions, as it gives you an idea which extensions are going to give you trouble. For extensions with qualify value greater than 300 ms you should definitely worry. For extensions at 2000ms delay or more, turning qualify off simply means to ignore the obvious problem. Such extensions have communication or network issues which require serious attention. You can set this parameter to, e.g. 3000 ms or more if dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall even at 2000 ms conversation is not truly real time and not easy. In our case the problem isn't that the phones are experiencing high latency per se but rather than a full pipe plus all these SIP messages is playing hell with the QOS stuff. 20 phones in one location times say 4 SIP packets every 2 seconds equals 40 SIP packets a second. That normally isn't a problem but when the pipe gets congested then we start seeing issues when a call comes in and 400 BLF notices go out etc. Obviously we can increase the amount of bandwidth reserved for SIP traffic but I'm just not sure why we're sending all those packets in the first place. In other words, the qualify traffic is actually causing the problem, not revealing it. Chris -- _ -- 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] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 12:42 PM, Leif Madsen wrote:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set(SIP/INTERTELin-, CDR(accountcode)=AZURAin) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set(SIP/INTERTELin-, BRON=473555006473555006) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto(SIP/INTERTELin-, vakantie) in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro(SIP/INTERTELin-, vakantie,58) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL(SIP/INTERTELin-, Connect connid localhost username
passwd AsteriskHosted) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL(SIP/INTERTELin-, Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58) in new stack
vps2301*CLI
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups
Dialplan :
[macro-vakantie]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten = s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,GoToIf($[${fetchid}==0]?exit)
exten = s,n,NoOp()
exten = s,n,GoToIfTime(${AST1}?opvakantie)
exten = s,n,GoToIfTime(${AST2}?opvakantie)
exten = s,n(exit),NoOp()
exten = s,n,Set(vakantieresult=continue)
exten = s,n,MacroExit
exten = s,n(opvakantie),NoOp(op vakantie !)
exten = s,n,GoToIf($[${NA}=hangup]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
I've seen problems with MYSQL() application crashing on customers boxes
before.
It is not that well supported, and would greatly recommend you move to
func_odbc
usage for dialplan-database integration.
Not only will it simplify your dialplan, but likely will resolve your crashing
issues as well. I've done this for at least 3 customers who were using MYSQL()
and all crashing issues stopped and their dialplans ended up becoming
significantly more readable.
This will also fix any real or perceived mysql takeover issues since
odbc can be attached to any backend without changing the code.
Leif.
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Re: [asterisk-users] Moving from DSL to T1
On 09/12/2010 02:34 PM, Kyle Kienapfel wrote: Really it depends on what the capabilies of dsl were assuming you are just using both dsl and t1 as internet connections. a dsl that has close to 1mb/sec out and 10mb/sec or so in, is going to be pretty comparable to a t1 actually so not really sure why you would make that switch in the first place. as long as there is a static ip for the server on either, you wouldn't see much difference. (t1 is actually usually delivered over hdsl which is basically the same thing as adsl except the bandwidth is more symetric.) if you have a low speed dsl, such as like 128kb/sec up and 512 down you'll see much faster performance, but again not much big diff if both are just internet connections. This is also assuming your carrier doesn't particularly grossly oversell either service. You need to make sure you are getting transit, not burstable, or quality may suffer depending on how its oversold. On Sun, Sep 12, 2010 at 10:43 AM, Richard Stuppi rich...@stuppi.com mailto:rich...@stuppi.com wrote: I work in a small office and have fallen into the role of network support based on knowing enough about networking to be dangerous. Our office is moving from DSL to a T1. Were using Asterisk as our PBX and I'm looking for hints or resources that might help me make the transition as error free as possible. Are there well known gotchas that I shoud be aware of? Thanks in advance, Richard Stuppi rich...@stuppi.com mailto:rich...@stuppi.com 626-221-8010 You should be more specific, A)Are you switching from voip over DSL to voip over T1 B) ... or using the T1 for phones? C)Are you switching from analog lines + DSL to just a T1 for voice and data? -- _ -- 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] spam blacklist
SIP wrote: what can you do ? simple discard spam don't bounce it. On 7/28/10 9:45 PM, Sam wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Bit of a pain in the left ventricle, really, but what can you do. N. -- _ -- 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] Foip solution
Mike Diehl wrote: Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from their computers? TIA, hylafax is the standard never had a problem with it. used to have the odd issue with a faxmodem on a fxs port from a channel bank, now have it on a virtual iaxmodem, no problems at all. In fact we have a whole bank of virtual modems and they work just fine. -- _ -- 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] identify the costumer
Douglas Pasqua wrote: Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify the customer's name in the table CDR. throw the information into a database and lookup the numbers before displaying the output with some other app. once they are in the db great but to build that up you could use any of the reverse lookup services and then just ask whoever made the calls to fill in whatever else is missing until you are up to a high hit rate of matches in your database. Have a procedure to enter new numbers for new clients. if you use something like sugar, just use the data in there to do your lookups since that is likely where the staff are looking up the numbers to make the calls in the first place. Thanks, Douglas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
Mark Willis wrote: This could potentially create a very weird audio situation where the delay between adjacent phones is audible so instead of acting like loudspeakers in parallel on a conventional system, it just sounds like a bunch of people talking at once and is not understandable. Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use a BLF for monitoring
Richard Kenner wrote: Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. That's what the Custom device type is for. please elaborate I would like to know too -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxmodem / hylafax receive problem
Kingsley Tart wrote: try several fax machines and see if you get the same results Hi, I'm trying to receive faxes using hylafax / iaxmodem but I just can't get it to work. We're using Sangoma E1 cards and have calls coming in over PSTN. I've tried turning hardware echo cancellation off but it makes no difference. This is what I get in /var/spool/hylafax/log: [r...@faxhost log]# cat c3 Jan 14 12:44:43.39: [ 3403]: SESSION BEGIN 3 18005551212 Jan 14 12:44:43.39: [ 3403]: HylaFAX (tm) Version 4.4.6 Jan 14 12:44:43.39: [ 3403]: CallID: 1234567890 /var/spool/asterisk/fax/20100114124443-1234567890-08451234567 NONE 08451234567 Jan 14 12:44:43.39: [ 3403]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Jan 14 12:44:43.39: [ 3403]: -- [4:ATA\r] Jan 14 12:44:47.11: [ 3403]: -- [7:CONNECT] Jan 14 12:44:47.11: [ 3403]: ANSWER: FAX CONNECTION DEVICE '/dev/ttyIAX0' Jan 14 12:44:47.11: [ 3403]: STATE CHANGE: ANSWERING - RECEIVING Jan 14 12:44:47.11: [ 3403]: RECV FAX: begin Jan 14 12:44:47.12: [ 3403]: -- HDLC32:FF C0 04 B5 00 AA 12 9E 36 86 62 82 1A 04 14 2E B6 94 04 6A A6 4E CE 96 F6 76 04 2C 74 2C 74 6C Jan 14 12:44:47.12: [ 3403]: -- data [32] Jan 14 12:44:47.12: [ 3403]: -- data [2] Jan 14 12:44:48.07: [ 3403]: -- [7:CONNECT] Jan 14 12:44:48.07: [ 3403]: -- HDLC23:FF C0 02 B6 A6 26 F6 B6 1A 82 92 04 04 04 04 04 04 04 04 04 04 04 04 Jan 14 12:44:48.07: [ 3403]: -- data [23] Jan 14 12:44:48.07: [ 3403]: -- data [2] Jan 14 12:44:48.81: [ 3403]: -- [7:CONNECT] Jan 14 12:44:48.81: [ 3403]: -- HDLC13:FF C8 01 00 77 5F 23 01 FB C1 01 01 18 Jan 14 12:44:48.81: [ 3403]: -- data [13] Jan 14 12:44:48.81: [ 3403]: -- data [2] Jan 14 12:44:49.39: [ 3403]: -- [2:OK] Jan 14 12:44:49.39: [ 3403]: -- [9:AT+FRH=3\r] Jan 14 12:44:56.39: [ 3403]: -- [0:] Jan 14 12:44:56.39: [ 3403]: MODEM Empty line Jan 14 12:44:56.39: [ 3403]: MODEM TIMEOUT: waiting for v.21 carrier Jan 14 12:44:56.39: [ 3403]: -- data [1] Jan 14 12:44:56.39: [ 3403]: -- [2:OK] Jan 14 12:44:56.39: [ 3403]: -- [9:AT+FRS=7\r] Jan 14 12:45:26.39: [ 3403]: MODEM TIMEOUT: reading line from modem Jan 14 12:45:26.39: [ 3403]: MODEM Timeout Jan 14 12:45:26.39: [ 3403]: Failure to receive silence (synchronization failure). Jan 14 12:45:26.39: [ 3403]: -- data [1] Jan 14 12:45:26.41: [ 3403]: -- [2:OK] Jan 14 12:45:26.41: [ 3403]: RECV FAX: Failure to receive silence (synchronization failure). Jan 14 12:45:26.41: [ 3403]: RECV FAX: end Jan 14 12:45:26.41: [ 3403]: Failure to receive silence (synchronization failure). Jan 14 12:45:26.41: [ 3403]: SESSION END I've tried starting hylafax with -o instead of -i but that makes no difference either. I previously had hylafax-6.0.4 and iaxmodem-1.2.0 but was getting the same problem. Any ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canadian call quality issue
Kyle Kienapfel wrote: Going along the internet between us and canada doesn't add much distance, but bouncing back and forth between east and west coast does. I'm not so sure that is the case, what I do know is both Rogers and Shaw can never seem to fix complaint issues with voip unless you are using their phone service. We just gave up on it and I will not ever spend a penny with Rogers as a result since I am convinced they are deliberately filtering things so you are locked into their voice services. Other than that, voip works just fine. On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote: hello, we have been using a couple of US based VoIP providers for outbound calls completed within the US, without any issues. We recently started making calls to Canada and have received a few complaints about the call quality. Questions : - Could this be because of the number of intermediate IP hops between us / our VoIP provider and the Canadian phone companies ? - Would choosing a Canadian VoIP provider address / resolve this issue ? Thank you in advance. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live CD - do you think they are worth doing?
Randy R wrote: I might try a live cd once or twice, or use it to boot a dead computer or one that is not mine, BUT for anything with any sort of time investment in settings to try anything you lose it all with a live cd so why bother since if you can't try it all in one session, you have to start over. Live usb sticks are another matter (assuming your bios actually reliably boots them) at least you can save your changes and pickup where you left off the next time. Hi, Curious, do many of you check out software or projects when they have a live CD or does that make any difference to you? Does anyone know if the general public (not reading this kind of list) is attracted to a Live CD more than an Install one? thx, /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
Travis Elsberry wrote: Hello all, Do you know if it IS possible to use multiple lines/extensions on SIP with a Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but have it register to a couple of different extensions, then use different ringtones to identify which line was ringing when a call came in. The grandstream gxp2000's have 4 lines, and this is what I did - I have 2 sip channels registered with the pbx, one for personal calls and one for business calls, so I can tell by which rings what type call it is, however what I did was use the 1st and 3rd channels since if I am on the line and another call comes in it rolls into the next line on the phone that way the roll happens to lines 2 and 4 that are not bound to an extension. if the roll happened to a channel that was registering rather than an empty button it would be confusing on the type of call. what is even more confusing is the 4 lines also roll into the speed dial / status buttons so if you have people on hold, it may look like you are getting a call from a speed dial link when its really an outside call. 4-6 lines sounds like a lot but I think in practice its more used for rollover than that many unique extensions on a phone. Thanks, Travis - Original Message - From: Michiel van Baak mich...@vanbaak.info To: asterisk-users@lists.digium.com Sent: Wednesday, November 25, 2009 2:40:07 AM Subject: Re: [asterisk-users] How many lines do you use. On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
Dean Collins wrote: Earlier in the thread someone made a comment about using gsm since everyone had gsm handsets already. Can you explain in detail please ? (what hardware specifically, and how does this actually work ?) My ignorant assumption is something like the end user has a cell phone that actually works with 2 carriers - yours and the real carrier. DECT rocks – I understand the reasons for wanting to use wifi but sometimes when it’s raining it makes more sense to drive a motorcar instead of ride a motorcycle J Cheers, Dean *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jason Baker *Sent:* Thursday, September 24, 2009 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP/WiFi handsets? I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. *Jason Baker */IT Coordinator/ *Glastender, Inc.* 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com/ mgra...@mstvp.com mailto:mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com mailto:jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] dCAP Exam
Benny Amorsen wrote: Jared Smith jsm...@digium.com writes: Not that I would ever consider taking an exam like that, but I have been using/configuring asterisk since nearly the beginning of this mailing list, and I have never touched dahdi or polycom. Someone should still be able to pass an exam without knowing about specific hardware where there is more than one alternative to use in real configurations. Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. I believe I can reveal this much without causing any problems for Digium: Be sure you have tried to configure a Polycom phone and an analog DAHDI card. Wasting 30 minutes on those two things makes passing the exam slightly more challenging... /Benny (whose only experience with analog DAHDI so far has been that dCAP exam) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com outbound mail slow?
Tony Mountifield wrote: I posted a message to this list about 50 minutes ago. I received a posting acknowledgement pretty quickly, and it showed up in the mailman list archives, but I still have not received a copy. Looking at some of the other recent messages I have received, they have also suffered a delay of 20 minutes or more on lists.digium.com before being sent out again. Are there any plans to beef up the mailing list server so that messages can get through with less of a delay? Cheers Tony uncheck your request confirmation setting in your mail client and think of the load that will save if everyone did that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Twitter is Suing me!!!
D. Dante Lorenso wrote: part of this is making a statement to get publicity, if twitter really didn't like what you were doing they'd simply cut off your app accessing their servers. But obviously that is not what its about. Dean Collins wrote: I received this email 30 minutes ago stating that Twitter is suing me?? Basically they feel that my application - www.MyTwitterButler.com http://www.mytwitterbutler.com/ does the following. *1/ That anyone using the API to auto follow people are breaching the TOS??* *2/ That no one can use the word “Twitter” in their domain* *3/ That somehow people might be confused my application is related to twitter even though every page is labeled //* I'd have to side with Twitter on this one: 1) From your domain name, I don't know that you aren't part of Twitter(tm) 2) Your service seems to be trying to circumvent what they are doing rather than just adding value to it. 3) You are using a Twitter-like bird in your logo at the top. I bet if you changed your domain name, changed your service name, created a different logo, and stopped infringing on their trade mark you'd be fine. I'll bet the Twitter TOS grants you rights to their API as long as you are only adding value and not trying to lead customers away from Twitter. They have the right to put something like that in there TOS. Then again, your app seems to heavily hit the Twitter API with 20,000 API calls an hour? I bet you don't pay monies to Twitter for all that system usage yet you are making money from sales of your app at $10 per license. If you sell 10,000 copies of your code, Twitter has to suffer through 20 million API calls per hour (480 BILLION calls per day) and you get to have $100,000. That doesn't seem fair does it? -- Dante -- D. Dante Lorenso da...@larkspark.com 972-333-4139 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Carlos Ruiz Diaz wrote: I did not know that the price was that low. Anyway, for people living really far from USA the price gets incremented twice or more and this is without considering the conversion between currencies. 1 $ = 5100 Gs., not cheap at all. all that stuff is coming from hongkong and china - tons on ebay and auctions in all sorts of currencies with nearly free shipping, so yes it is cheap, does not matter where you live, you just need to look. Thanks. On Sun, Aug 2, 2009 at 3:07 PM, jon pounder j...@inline.net mailto:j...@inline.net wrote: Carlos Ruiz Diaz wrote: Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. the digium single fxo cards and clones for about $10 ARE modems. you can get a sip gateway fxo + fxs in one box for about $50 really - how much cheaper do you want ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can 2 quad T1 cards work in 1 quad core amd server
Jerry Geis wrote: oh you mean a telemarketing pest server ? I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can 2 quad T1 cards work in 1 quad core amd server
David Backeberg wrote: On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote: I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Yes. Buy a server that has the corresponding ports to accommodate the cards. A modern server is probably going to have PCI-E slots and you'll want the appropriate TDM cards. Any thoughts. Yes. That's a lot of power to drive a comparatively small number of calls. Also, I find it interesting that so many of the answers to these questions turn into a: 'you're going to use that for bad purposes' which is retorted with: 'no I'm not' well just look at the fact that the cost to place a call is declining and the cost to mail a letter is increasing, then you see where the focus of abuse is going. I will say that I make a boatload of these outgoing 'play a file' calls, and they are for legitimate purposes to existing customers. Is it really that hard to imagine a business with a good reason to call their customers? I think that if somebody asked for a postal machine that processed a large number of letters somebody would say 'you're using that for junk mail', and somebody else would say 'fine, I won't mail you your paycheck'. I mostly just think it amusing that everybody has bad motives until proven otherwise. And moreover, the idea that you can somehow inoculate the world from people with bad motives if you don't provide assistance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Miguel Molina wrote: randulo escribió: On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote: I run my analog telco over cat5, but that's in-house and definitely not 3km. That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) What I think about this is, the length of the copper cable between the central office and home is usually several km, but definitely helped by the central office circuitry (current source instead of voltage source, that guarantees a minimum ringing voltage on the far end). What I don't know is, a FXS port behaves the same as a central office, electrically speaking? If that is so, you could extend your 3km of cable without problems, but I think you can have some noise problems depending on what places the cable has to go through. from the ringing point of view, the CO ring generator is usually truly a sine wave and this propagates well through a cable. The cheap fxs ports are mostly square waves and lower voltages with limited current sourcing (check the REN numbers they are capable of ringing for a comparison if its listed) some of the cheap ones have trouble ringing a phone plugged in with a short line cord. So in addition to being frequency choked by the long run the square wave will get reduced in amplitude, and it may well have been marginal amplitude to begin with. so depending what fxs hardware you have driving it and the load from the phone, results will range from works perfectly to does not work at all. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Wilton Helm wrote: one thing I missed mentioning about fxs devices - the linksys/sipura ones actually allow you to set line characteristics on the slic inside it. you can vary from the 600ohm default, and tweak gains a bit. Some mix of a capacitive line or different resistance may help. never tried myself but there are a ton of things you can play with. There are a lot of factors that impact this. First, CAT 5, while usable is overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice grade lines. That being said, for that long a run, a heavier gauge wire would be better. I believe telcos use 18 – 22 guage (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss. Most FXS or ATA devices use 24 volts or less for “battery”. That works fine for short loops, but limits the range. A central office POTS port normally uses 48 VDC which works well to several KM. If the customer is at the end of a long run in a rural area, they use a “long line” card which uses 75 volts. (In rural communities, they often place the line cards in a roadside “remote terminal” and use statistically multiplexed T1s to make it appear to the switch as a part of it. That addresses the DC characteristics, which can be reduced to ohms law. A phone needs around 8 V @ .02 A. The wire resistance determine the drop (E = IR) and the source voltage determines whether there will be enough left. The A.C. characteristics are more complicated. The FXS must do a 2 wire to 4 wire conversion, which involves matching the impedance of the line. The FXS is generally designed for relatively short lines, so might not be able to match either the resistance or capacitance found in a long run. Heavier wire will minimize this. In addition to that, the transmit side of the 2 wire to 4 wire circuit must be able to drive the load it sees, and again it may not be designed with a long run in mind. Finally, COs line cards have the ability to adjust receive and transmit gain to compensate for sound level losses in long lines. While this isn’t routinely done on simple circuits, it is an option an FXS doesn’t generally have. In addition, the more gain that is inserted, the harder it is to balance to 2 wire to 4 wire circuit, and the more complex it has to be in order to support this. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
Diogo Saad wrote: how do I configure my SIP account information? I mean, sip proxy and etc. you need just a couple pieces of information server (put this in any setting that says proxy or host etc, all set the same) account (the extension in asterisk, put anywhere that sounds like a non-display only field) password (secret, key, password etc., should be one field that takes this in the config) register = yes basically that's it. you mean need to disable feature codes etc, but the above will get most any sip device working with asterisk once you setup an extension for it. On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net mailto:j...@inline.net wrote: Diogo Saad wrote: Using an ATA, Do I still need a softphone or it´s embedded in the hardware? plain old walmart phone plugs in the ata (with or without callerid, adsi, cordless, etc) On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org http://asterisk.org http://asterisk.org@sedwards.com http://sedwards.com http://sedwards.com wrote: On Tue, 26 May 2009, Diogo Saad wrote: What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? An Ethernet based ATA would be more versatile. I like Digium's discontinued IAXy. Dead simple to configure, easy to travel with, no NAT headaches. Used on ebay should set you back about US$30. Thanks in advance, Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
Wilton Helm wrote: You are exactly right. Cat 5 had no advantage over cheaper wire for voice, and the length limitations are meaningless. Consider that Cat 5 is typically use with signals that extent to 30 MHz or beyond. A voice grade analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, the wire generally doesn’t even act like a controlled impedance. I completely agree, but that said I still use cat5/e for everything anyway, not worth having more than one kind of wire, and lets you change your mind later on the usage. Once you are using outside plant facilities though, you live with what you get, and don't expect much. Wilton *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Tuesday, May 26, 2009 8:50 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Maximum cable length for analog phone from FXS port I could be wrong but I don’t think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
asterisk-us...@rogg.is wrote: couple last words on this - if that is the application, then ringing the remote terminal may not even be necessary, you really only care about the hookswitch and audio which is a different thing entirely from ringing. You may be able to boost the battery voltage with a simple dc adapter in series to get the line build out capability you need. Just make sure its floating with respect to ground and wire it in. Don't be afraid of hurting the phone, you won't. Check out dialplans for the ATA's for warmline and hotline for emergency phones you will probably want this. This is one I use for example warm dial - wait 3 sec and if not 3 digit number dial 100 ( P3 :100|xxx ) ie: if they know who they want locally on the pbx they have 3 seconds to dial it, otherwise the ata just dials extension 100 which is an entrypoint to one of the IVR trees on my system. Appreciate all your input folks. Much of it very helpful in the greater context of the initial question. Thank you for the suggestion of using various wireless devices, but I'm stuck with fixed wiring since this is a security/emergency phone(s) installation underground in large tunnels. Also, switching to VOIP is not really the answer here because then I'm forced to solve a lot of power, repeaters/switches problems that arise. So I'm actually worse of than using the analog connections I think. I do have some control over the wiring/cable chosen for this project but still forced to find a solution where I can feed the analog phone line the total 3km line distance. I would love to find a way to do this in the Asterisk context with some sort of FXS feed, either from Digium (or compatible) hardware or any of the available ATA boxes. The Sapura box suggestion may be something and I'll look closer into that as well as continuing to look for other ways to do this. tnx! Baldvin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 26. maí 2009 19:42 To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I would suggest making a wifi connection with directional hi-gain antenna's. Ans a small box at the other end. Have a look at: http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit- pc.info/downloads/handleidingen/fit_pc_2_eng.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
John Novack wrote: If this is an emergency phone situation then I would question the wisdom of even considering using Asterisk. Conventional telephony solutions exist that will easily cover the loop length and provide the reliability that should be required by risk management in such a situation. why are you going on the assumption asterisk is somehow inherently less reliable than a conventional solution ? I am not trying to start any sort of war here, but is that based on any sort of facts ? hardware wise its basically all the same electronics whether they were meant as a general purpose computer or a telephony specific computer - they all fail eventually and the MTBF is usually related to the relative price in the specific market. I have not really had any software reliability problems in years of running asterisk (although some do and I am sure there are firmware revs for pbx's that have issues too) so why make that general statement ? as far as risk management - any one system can fail, end of story. Risk management would entail a backup system if failure of the primary is not acceptable. In a tunnel application physical damage to the wiring is probably a lot more likely than a hardware failure, be it from accident, fire, collapse etc., meaning when you need the phone most, it is least likely to work. Those factors would affect any hardwired telephony solution equally. John Novack asterisk-us...@rogg.is wrote: Appreciate all your input folks. Much of it very helpful in the greater context of the initial question. Thank you for the suggestion of using various wireless devices, but I'm stuck with fixed wiring since this is a security/emergency phone(s) installation underground in large tunnels. Also, switching to VOIP is not really the answer here because then I'm forced to solve a lot of power, repeaters/switches problems that arise. So I'm actually worse of than using the analog connections I think. I do have some control over the wiring/cable chosen for this project but still forced to find a solution where I can feed the analog phone line the total 3km line distance. I would love to find a way to do this in the Asterisk context with some sort of FXS feed, either from Digium (or compatible) hardware or any of the available ATA boxes. The Sapura box suggestion may be something and I'll look closer into that as well as continuing to look for other ways to do this. tnx! Baldvin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 26. maí 2009 19:42 To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I would suggest making a wifi connection with directional hi-gain antenna's. Ans a small box at the other end. Have a look at: http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit- pc.info/downloads/handleidingen/fit_pc_2_eng.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
Jonathan Moore wrote: On Wed, May 6, 2009 at 10:53 PM, John Novack jnov...@stromberg-carlson.org wrote: Not sure how you would do that, as the X100 card is an FXO card, won't provide either battery or dial tone to the cordless. What you will want for that is an FXS card or ATA. The X100 card will connect to a central office line, and with the later software echo cancel works OK. Not nearly as bad as some have made it out to be, though for US/Canada lines. Not suitable for UK and others yeah I agree with the above - I never really found echo to ever be a problem, my only complaint was on some less than stellar cpu's I was having dtmf recognition problems. Ah, yes. Thanks for correcting me on that, I was getting some things mixed up in my head. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wireless ATA
Jeff LaCoursiere wrote: why not just put something like a wet11(wireless bridge) and pap2t(2x fxs) in the same box ? dev time = 0 cost ~100 those are just two of the many products that would work together to do what you want. I have a need for an ATA that will register over wifi. *NOT* a DECT phone or other cordless type phone plugged into a wired ATA. Not seeing one right off, and following the recent discussions about compact fanless systems, I thought a custom build-your-own might not only be useful for my purpose, but may be a viable product in its own right. So I started looking for mini-PCI based FXS cards. The only one I seem to find is the OpenVOX A400M. Anyone have experience with it? Any alternatives? Having perused the last fifty or so emails on the various motherboards available, several seem to have two mini-PCI ports, so the very basic idea is to buy one of them and put both an OpenVOX card and a 802.11g radio card in it. Is there any reason these smallish motherboards would be unable to power both boards? Heat issues? Any suggestions on how to choose one of the wifi cards? I would like to run DD-WRT on this beast to handle the wifi-client mode, and I noticed (though have not tried) that DD-WRT has a build that includes asterisk, which would of course drive the FXS port. I figure the whole thing should cost less than US$250, and in quantity perhaps much less. Another thought is to skip the whole FXS port card and use the built-in sound interface to the small motherboard to drive a simple headset based phone. That would be significantly cheaper. If anyone is as interested as I am in creating this beast and would like to donate the needed pieces to the cause, I will commit to writing a web interface and the pieces to do remote provisioning, and we can market the new product... Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wireless ATA
Jeff LaCoursiere wrote: On Mon, 4 May 2009, Jon Pounder wrote: Jeff LaCoursiere wrote: why not just put something like a wet11(wireless bridge) and pap2t(2x fxs) in the same box ? dev time = 0 cost ~100 those are just two of the many products that would work together to do what you want. But that wouldn't be anywhere near as fun :) It doesn't accomplish the end goal of having a product, which I think there would be a market for, either. The cost is a bit more than $100, though, too - WET11's seem to go for about $70, and the PAP2T for $50... but that is still significantly less than what I am talking about. Would anyone buy such a product if it existed? Personally I would go for the approach of having a couple off the shelf products even if they were inside a box to pretty it up. That way I know when there is a problem, I can just run out to any big box store and get parts to fix it quickly, and if I am keeping spares around the wired/wireless spares are partially the same components. I tend to stay away from specialty devices where possible just because when you need a replacement NOW its hard to find one. j I have a need for an ATA that will register over wifi. *NOT* a DECT phone or other cordless type phone plugged into a wired ATA. Not seeing one right off, and following the recent discussions about compact fanless systems, I thought a custom build-your-own might not only be useful for my purpose, but may be a viable product in its own right. So I started looking for mini-PCI based FXS cards. The only one I seem to find is the OpenVOX A400M. Anyone have experience with it? Any alternatives? Having perused the last fifty or so emails on the various motherboards available, several seem to have two mini-PCI ports, so the very basic idea is to buy one of them and put both an OpenVOX card and a 802.11g radio card in it. Is there any reason these smallish motherboards would be unable to power both boards? Heat issues? Any suggestions on how to choose one of the wifi cards? I would like to run DD-WRT on this beast to handle the wifi-client mode, and I noticed (though have not tried) that DD-WRT has a build that includes asterisk, which would of course drive the FXS port. I figure the whole thing should cost less than US$250, and in quantity perhaps much less. Another thought is to skip the whole FXS port card and use the built-in sound interface to the small motherboard to drive a simple headset based phone. That would be significantly cheaper. If anyone is as interested as I am in creating this beast and would like to donate the needed pieces to the cause, I will commit to writing a web interface and the pieces to do remote provisioning, and we can market the new product... Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] want to set up text based adventure for asterisk
Eric Fort wrote: Anyone know where I could find a good beginning for using asterisk and the text based game adventure together such that I could play from the nearest phone? google on collossal cave. honestly its the absolute worst unreadable mess of code ever conceived by man or beast. that said I made a web version of it that actually works here is where you start - hack the only semi unreadable portion for save and load games so it does not cost you points to save and load. then make a script that does this : start game, load game, inject next command to game, trap output of that move, save game again every time the user does a command, run the script and then show/speak them the output. yes its a kludge of the Nth order but at the end of the day it works, and you didn't even have to understand the garbage code that drives the thing. let me know how it goes. Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX IP Phone - PAP2Ts
Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, bilal ghayyad wrote: on the subject - I have one and it runs VERY hot - is this normal ? I am almost afraid to leave it on its so hot. Thanks a lot, but I need IAX IP PHone. About the PAP2Ts, in which price u r getting it? Any link that u can advise me to check it? Regards Bilal Just plug it into google. You will be awash in people trying to sell you PAP2Ts. I'm buying from IP Phone Warehouse... I think the going price is around US $50. Cheers, j --- On Fri, 4/17/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: IAX IP Phone - PAP2Ts To: bilal ghayyad bilmar...@yahoo.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, April 17, 2009, 12:50 PM On Fri, 17 Apr 2009, bilal ghayyad wrote: Dear Jeff; PAP2Ts support IAX? And support G729 codec? And it support ddns (or dns) to be used instead of the IP Address? Regards Bilal Hi Bilal, No, the PAP2Ts do not support IAX. They do support G.729a and DNS however. Cheers, j --- On Mon, 1/19/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] IAX IP Phone To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: bilmar...@yahoo.com Date: Monday, January 19, 2009, 12:58 PM On Mon, 19 Jan 2009, Joseph wrote: On Mon, 19 Jan 2009, bilal ghayyad wrote: Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal How about IAX2 adapter from digium? I've been uing it and it works very well. Wow, that has NOT been my experience, though it has been a few years (2005) since I used them. The ones I purchased were first of all expensive. They overheated and froze up often. Only a single port. No dual ethernet option. Provisioning is a PITA. Codec support was minimal. I thought at the time that being IAX I wouldn't have to worry about NAT issues and that was worth the extra difficulties, but I have been using Linksys PAP2Ts ever since and never looked back. And it has been over a year since I had any NAT issue to deal with, though have now installed them in hundreds of different configurations. Perhaps these things have been rectified since... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX IP Phone - PAP2Ts
Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jon Pounder wrote: Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, bilal ghayyad wrote: on the subject - I have one and it runs VERY hot - is this normal ? I am almost afraid to leave it on its so hot. That doesn't sound right. One sitting next to me is used quite often and it is only a bit warm. Certainly not hot. ok thanks, likely it has sort of problem, I have other linksys devices in the same family and none run nearly that hot. j Thanks a lot, but I need IAX IP PHone. About the PAP2Ts, in which price u r getting it? Any link that u can advise me to check it? Regards Bilal Just plug it into google. You will be awash in people trying to sell you PAP2Ts. I'm buying from IP Phone Warehouse... I think the going price is around US $50. Cheers, j --- On Fri, 4/17/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: IAX IP Phone - PAP2Ts To: bilal ghayyad bilmar...@yahoo.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, April 17, 2009, 12:50 PM On Fri, 17 Apr 2009, bilal ghayyad wrote: Dear Jeff; PAP2Ts support IAX? And support G729 codec? And it support ddns (or dns) to be used instead of the IP Address? Regards Bilal Hi Bilal, No, the PAP2Ts do not support IAX. They do support G.729a and DNS however. Cheers, j --- On Mon, 1/19/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] IAX IP Phone To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: bilmar...@yahoo.com Date: Monday, January 19, 2009, 12:58 PM On Mon, 19 Jan 2009, Joseph wrote: On Mon, 19 Jan 2009, bilal ghayyad wrote: Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal How about IAX2 adapter from digium? I've been uing it and it works very well. Wow, that has NOT been my experience, though it has been a few years (2005) since I used them. The ones I purchased were first of all expensive. They overheated and froze up often. Only a single port. No dual ethernet option. Provisioning is a PITA. Codec support was minimal. I thought at the time that being IAX I wouldn't have to worry about NAT issues and that was worth the extra difficulties, but I have been using Linksys PAP2Ts ever since and never looked back. And it has been over a year since I had any NAT issue to deal with, though have now installed them in hundreds of different configurations. Perhaps these things have been rectified since... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
Erick, how about posting your home phone number here so we can all call you and play a 20second audio clip - I am sure you would see nothing wrong with that would you ? ContactTel Business wrote: Your right, i don't think we would help someone asking on advice to send 1 million emails for Viagra would we ? So why the hell aren't we thinking straight and tell the poor guy? Ive seen dialer app that where legit, even worked on some for the military. But this is just spam /pham (phone spam) send 10USD to my email ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-02-09 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
Roger Marquis wrote: Steve Totaro wrote: I understand you are a developer and you want IAX2 to be great. That is your job, but the fact is that it is not and has caused audio and security problems for YEARS in EVERY release. It should bug you and everyone at Digium that waves the IAX2 flag. Can you elaborate on these audio and security problems Steve? Looking at the two protocol specs I cannot see a basis for your claim. IAX doesn't embed the local IP address in the packet data but that's surely no substantive security. It does separate data and signaling at the application-level, but again, that's no basis for such a claim. Protocols must be looked at separately from their implementations. From the various responses it appears that Asterisk 1.4's implementation of IAX has flaws. These do not necessarily reflect on the protocol. OTOH, there are a lot of engineers with SIP skill and experience who, naturally, are concerned with their investment in time, education, and experience. While this may or may not apply to Sonicwall engineering, it's also true that any streaming protocol will be better handled by devices that process packets in ASICs (high-end firewalls) rather than CPUs (PCs and low-end firewalls). This sounds like a bunch of gobbledegook spewed out by those very high end firewall vendors. Call it what you want but anything that processes packets in any way and makes a decision on what to do is by definition a CPU. And a general purpose CPU is not exactly poor at the job. If you look at utilization levels and latency on a typical CPU you would have thrown away already as a server, its barely even noticable utilization running a complex set of rules on a high volume data stream. FWIW (2 data points) I get uniformly better service from our IAX trunk provider than our SIP trunk provider. No idea whether that's protocol, implementation (1.4 on my side), or provider-related though I suspect the later. Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
David Ruggles wrote: I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? I just use the web interface, and note that it only works when you have connected networks, either local or on a vpn. There is an endpoint manager in trixbox that will find devices but only on the local lan pretty much. I generally configure the phones how I want at my desk, and then give them to whoever they are for and they plug them in remotely someplace and they work as per setup. My setup is a mix of grandstreams and other zap and sip devices, but I can imagine doing updates on hundreds of phones would need to be more automated. Not sure if this exists or not but since its only form posts to configure the phone, something that wrote out files, from a database and used wget or curl to dump them into a phone would be pretty useful, but only if the phones http is accessible from the server, and only if you had a lot with similar configurations you could template. TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gpx 2000 Busy Lamp Field
Vieri wrote: I see much the same except I think if you investigate further, the light will be green whether the phone ever registered or not. --- On Tue, 3/24/09, Ken Williams k...@intermountainelectronics.com wrote: Our work around is to lower the registration expiration on the phones. Well, something's not working as I expect it to. My GXP2000 phones have re-registration timeout of 2 minutes. In my example below, extension 4061 is unplugged (so obviously it won't re-register, ever). Meanwhile, the other GXP2000 phones keep seeing 4061 as on-line (green led). This happens with Asterisk 1.4.24. If I use Asterisk 1.2, the green LED turns red within a few seconds. My guess is that Asterisk is not sending the device state, or it's not correctly detecting that the phone has not re-registered. I might not be interpreting the hints logic correctly, but I suppose that if a sip show peer reveals that the extension is not registered (when it was before) then Asterisk should send out a message to all subscribed phones so that their BLF can be updated. As you can see below, 4061 is switched off, isn't physically online anymore, but all BLF LEDs are green. Why is this? # asterisk -rx sip show peers | grep 4061 4061/4061 (Unspecified)D N A 0UNKNOWN # asterisk -rx sip show peer 4061 * Name : 4061 Secret : Set MD5Secret: Not set Context : from-internal Subscr.Cont. : Not set Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 2 Pickupgroup : 2 Mailbox : 4...@device VM Extension : *97 LastMsgsSent : 0/0 Call limit : 50 Dynamic : Yes Callerid : device 4061 MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : Always ACL : Yes T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : (Unspecified) Port 0 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 4061 SIP Options : (none) Codecs : 0x2 (gsm) Codec Order : (gsm:20) Auto-Framing: No Status : UNKNOWN Useragent: Grandstream GXP2000 1.1.6.46 Reg. Contact : sip:4...@10.215.146.161:5060;transport=udp # asterisk -rx show hints | grep 4061 4...@ext-local : SIP/4061Custom:DND4 State:IdleWatchers 1 *764...@ext-dnd-hints : Custom:DEVDND4061 State:IdleWatchers 0 *214...@ext-findmefollow: Custom:FOLLOWME4061 State:IdleWatchers 0 PS: (I don't understand why 4...@ext-local has State:Idle) Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conference and wifi phones
Frank Bulk wrote: In a SOHO environment I would agree with you, but not if your coverage area needs to be tens of thousands of square feet. Deploying a complete overlay wireless infrastructure doesn't make sense and is another infrastructure to manage and maintain. did you think about your numbers before posting this ? what access point does not have a 50ft radius ? 100x100 ft is 1square feet = a single AP for 10's of 1000's of square feet, hardly a huge undertaking. Dect is not going to let you roam into another network or hotspot and still work, nor is it going to support the myriad of other devices that work on wifi networks. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo Sent: Tuesday, March 24, 2009 1:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] conference and wifi phones snip I have never understood why anyone would use wifi just to get cordless facility when DECT works so much better. snip /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
