Re: [asterisk-users] Panasonic PBX connect to Asterisk

Use a sip to PRI gateway or a PRI card in the asterisk system. Connect that to the Panasonic TDA600 using a PRI card on the panasonic side (KX-TDA0290). this will be the most worry free solution. On Wed, Sep 14, 2016 at 2:05 AM, Harry McGregor wrote: > Hi, > > > You need

Re: [asterisk-users] What linux distro most popular for Asterisk

Slackware here. On Thu, Oct 17, 2013 at 8:57 PM, Tiago Geada tiago.ge...@gmail.com wrote: debian wheezy compiled asterisk from source On 18 October 2013 00:27, Andrew Furey andrew.fu...@gmail.com wrote: [Apologies, top-posting, Gmail, yadda yadda] As with a lot of software, I suspect

Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

There are some appliances that support it is well. But those don't have a scanner, just thru the computer. MultiTech FaxFinder comes to mind, for the price they are excellent. On Fri, Jul 6, 2012 at 3:13 AM, Olivier oza_4...@yahoo.fr wrote: 2012/7/5, C F shma...@gmail.com: I searched a bit more

Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

T.37 http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation) There were some scanners manufactured with this in mind, however I cant remember who made them. On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I'm curious about the availability of Multi Function Printers with

Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

I searched a bit more, http://www.muratec.com/catalog/F320_config.html#email The above model supports t.37 but no sure if you can have it function such that any number entered will actually be send to a gateway. On Thu, Jul 5, 2012 at 10:20 AM, C F shma...@gmail.com wrote: T.37 http

Re: [asterisk-users] Advertising oportunity.

asterisk biz On Tue, Jun 12, 2012 at 3:10 PM, Jonson Player jonsonpla...@gmail.com wrote: Hello, I don't know if this list is appropriated to this subject but I want to ask you if there's some list where I can make an advertising announce for a new sip web site that was just launched. hank

Re: [asterisk-users] Free calls to the uS question

Doesnt google voice offer that? On Fri, Mar 30, 2012 at 1:00 PM, Christian christia...@runbox.com wrote: Hi all, Does anyone know of any providor that offers free calling to the US? Feel free to contact me off list. Many thanks, Christian --

Re: [asterisk-users] Polycom firmware 4.0.1 and paging

Are you using the same cfg files? If yes I would try rewriting them from scratch using the blank cfg files that come with new firmware. I have seen wiered things by using olde cfg files On Friday, February 10, 2012, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from

Re: [asterisk-users] Is this doable?

On Wednesday, February 8, 2012, Josh mojo1...@privatedemail.net wrote: http://www.asterisk.org/astdocs/node66.html Thanks, never knew that! Yes, I understand that it's not what you want, but that doesn't make it a security concern. If Asterisk is publicly available on one interface, making

Re: [asterisk-users] Is this doable?

On Wednesday, February 8, 2012, Josh mojo1...@privatedemail.net wrote: I don't get this. Didnt EVERYONE know it's insecure? Can you read? Can everyone? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Router that support Asterisk

G Have you ever heard of Google? Here is a link on google: http://lmgtfy.com/?q=google On Wed, Feb 1, 2012 at 2:17 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I heard from some friends that there are a dsl router that has Linux OS and it has asterisk on it, so the ip phone can

Re: [asterisk-users] Is this doable?

Whats asterick? On Wed, Feb 1, 2012 at 7:48 PM, Josh mojo1...@privatedemail.net wrote: I am trying to configure Asterick, having the following system setup on the Asterick server: * eth0 faces the external Internet interface, *but* it does not have IP address (it has a private one given to

Re: [asterisk-users] externip nat audio sip trunk issue problem

On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: Hi all,   I've tried search this problem on the list... no luck...   The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the

Re: [asterisk-users] Cell Phone as a Queue member

Use local channel 2012/1/31 Niccolò Belli darkbas...@gmail.com: Hi, Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to call SIP/$TRUNK instead. Cheers, Darkbasic -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Real T1 trunk group...

On Mon, Jan 16, 2012 at 5:48 AM, Louis Carreiro carreir...@gmail.com wrote: Hey all! I'm not sure if this went out the first time I sent it so I apologize now if it's a duplicate. I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I

Re: [asterisk-users] Best non polycom SIP conference room phone

snom conference room phone? Oh. So its political? They still make the best quality phones. Bryant Zimmerman On Jan 8, 2012, at 10:59 AM, C F shma...@gmail.com wrote: I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality

Re: [asterisk-users] Best non polycom SIP conference room phone

On Sun, Jan 8, 2012 at 3:38 PM, Carlos Alvarez car...@televolve.com wrote: On Sun, Jan 8, 2012 at 9:02 AM, C F shma...@gmail.com wrote: I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality. We have no problem

Re: [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?

Exactly which IE message are you trying to push manually? you shouldn't have to do that, it should be done in the configs for you. On Mon, Jan 9, 2012 at 1:45 PM, Johann Steinwendtner steinwendt...@gmx.net wrote: On 2012-01-09 17:46, Alex Villací­s Lasso wrote: I am trying to collect

Re: [asterisk-users] Connecting to an Old Phone System

What type of phone system? And what type of connectivity are you trying to give the old pbx? On 1/6/12, Dan Journo d...@keshercommunications.com wrote: Hi, This is not strictly an asterisk questions, but... ive got a client with an old digital pbx phone systems connected to an isdn30e line.

Re: [asterisk-users] Best non polycom SIP conference room phone

I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality. However check out Panasonic. They make a sip conference phone. On 1/5/12, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com

Re: [asterisk-users] Best non polycom SIP conference room phone

On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com wrote: I am looking for a really good SIP conference room phone for use with asterisk. I do not like Polycom at all. You have a really bad taste. What would you all recommend? I have to be able to get them in the US. I

Re: [asterisk-users] A new hack?

On Tue, Dec 6, 2011 at 5:19 AM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote: snip Your security needs depends on your environment. At this point in time, all of the hosts I manage for my clients exist in very limited environments and have

Re: [asterisk-users] A new hack?

On Fri, Dec 2, 2011 at 11:35 AM, Jim Lucas li...@cmsws.com wrote: On 11/26/2011 5:00 PM, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from

Re: [asterisk-users] A new hack?

writing (by hand) iptable rules? On Mon, 5 Dec 2011, C F wrote: Sorry I wasnt very clear in my first writing, I'll try to clarify. Using iptables only detects one type of attack (aggressive connections). While his machines might be secure enough to allow any other attacks and still not compromise

Re: [asterisk-users] A new hack?

On Thu, Dec 1, 2011 at 8:15 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 29 Nov 2011, C F wrote: BTW, you were just proven wrong, you need it for this hack. In addition to the few hundred protected asterisk installations I run, I also run a few honeypots. Protected? You

Re: [asterisk-users] A new hack?

On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host

Re: [asterisk-users] A new hack?

On Mon, Nov 28, 2011 at 10:57 AM, Tom Browning ttbrown...@gmail.com wrote: On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson gordon+aster...@drogon.net wrote: Linux has excellent built-in subsystems to control firewalling and so on without resorting to external programs. It's called iptables.

Re: [asterisk-users] A new hack?

On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting.  And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful

Re: [asterisk-users] DID and how the caller id will appear

On Tue, Oct 18, 2011 at 8:46 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; By the way, the asterisk version that I have is 1.8.4.2 and DAHDI version is 2.4.1.2 Here I would like to mention the following: 1) As per the telecom provider, they said they openned for us all the digits

Re: [asterisk-users] Maybe slightly OT but..

On Mon, Oct 10, 2011 at 10:31 PM, linux guy linuxguy...@gmail.com wrote: On Mon, Oct 10, 2011 at 8:08 PM, Andres and...@telesip.net wrote: I would recommend Acrobits.  Not free but only a few bucks.  It works fine with ATT 3G. This begs the question... which is more expensive (and where)...

Re: [asterisk-users] Call does not pass through

Can you please post the relevant parts of extensions.conf? As well as a CLI output of when you dial and it fails? On 9/26/11, Malvin Rito mr...@mail.altcladding.com.ph wrote: Hi list, My call does not pass through on the first dial, I have to redial again the number for the call to pass

Re: [asterisk-users] DID and the Caller ID for outgoing

Did you add the Set(CALLERID(num) as I have previously pointed out? On 9/25/11, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has effecting on the DID and Caller ID to appear at the destination, because I found the

Re: [asterisk-users] DID and how the caller id will appear

Confirm with your provider that allow you to set caller id on outbound. On Sun, Sep 25, 2011 at 1:59 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; By the way, the asterisk version is: 1.8.4.2 Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no success. Also I

Re: [asterisk-users] DID and how the caller id will appear

Set(CALLERID(num)=5631040) add this before the Dial command. On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; The DID range that we took from the telecom starts from 1030 and end by 1059, now whenever we place a call, the destination see the number 5631030. I

Re: [asterisk-users] Problem with multiple sip-peers against the same host

Can you please post: 1. Relevant sip.conf 2. sip debug when trying to make a call? On Thu, Sep 22, 2011 at 7:26 PM, David Björkevik da...@dynamore.se wrote: Dear list, We are switching to a new provider for SIP-trunks. We have 20 numbers, each defined as a separate SIP peer. With the old

Re: [asterisk-users] Ghost DID in System

Without your dialplan there isnt much that can be done to help. Can you please post your relevant dialplans? Whats voip1 and voip2? When you say outside the voip system call goes thru, to where? Who has the number currently? Any sip debug you care sharing? On Mon, Sep 19, 2011 at 6:51 PM, Aaron

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

Why php? Isn't vi the only way? On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati virbh...@gmail.com wrote: Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files

Re: [asterisk-users] Faxes suddenly failing

I think you should change the subject line to: Faxes suddenly worked for 2 weeks. On Wed, Aug 31, 2011 at 3:49 PM, Tim King tim.compnetw...@gmail.com wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During

Re: [asterisk-users] Dragging the dialup customers along, possible?

On Wed, Aug 31, 2011 at 9:25 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 08/29/2011 10:32 PM, C F wrote: On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitchca...@usawide.net  wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] Avaya to Asterisk Voice mail

What exactly is your setup? On Tue, Aug 30, 2011 at 10:44 AM, Dustin fails wff...@gmail.com wrote: Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue line. The issue I am having is Avaya is sending the originating caller id not the station id so Asterisk see that

Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

like this did the trick. Whenever it resync profile, it reboots, so setting resync to an hour nobody uses it and unseting resync periodicaly solve it for me. Jose Flores Galicia BriefCode Code Based Training 2011/8/15 C F shma...@gmail.com On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian

Re: [asterisk-users] Dragging the dialup customers along, possible?

From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing

Re: [asterisk-users] Dragging the dialup customers along, possible?

On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitch ca...@usawide.net wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Monday, August 29, 2011 3:18 PM To: t...@ewebforce.net; Asterisk Users

Re: [asterisk-users] Looking for ideas for nice home phone system

The 824 is NOT discontinued. On 8/23/11, John Novack jnov...@stromberg-carlson.org wrote: C F wrote: On Tue, Aug 23, 2011 at 5:21 PM, John Novack jnov...@stromberg-carlson.org wrote: snip What do you mean by MD? MD is a common telephony term for Manufacture Discontinued John Novack

Re: [asterisk-users] Looking for ideas for nice home phone system

...@stromberg-carlson.org wrote: NEC-DX-40 is another best buy Single pair phones 2 analog station ports door box ports AND remote programming via LAN or WAN Voice mail available with or without email notification SIP gateway option Far superior to the TA-824 and Partners John Novack C F

Re: [asterisk-users] Looking for ideas for nice home phone system

On Tue, Aug 23, 2011 at 5:21 PM, John Novack jnov...@stromberg-carlson.org wrote: C F wrote: Everything you mention for the NEC system is available with the 824 but the Lan/Wan programming. In fact on the 824 every port (to a system max of 24) or analog as well as proprietary. Support

Re: [asterisk-users] Looking for ideas for nice home phone system

On Tue, Aug 23, 2011 at 6:59 PM, Linuxguy123 linuxguy...@gmail.com wrote: On Mon, 2011-08-22 at 22:13 -0400, C F wrote: Panasonic KX-TA824 Or the Panasonic KX-TAW848 Or the Avaya Partner ACS 8.0 Are these Asterisk/VOIP based solutions ? I tried using google translate to see

Re: [asterisk-users] Looking for ideas for nice home phone system

Panasonic KX-TA824 Or the Panasonic KX-TAW848 Or the Avaya Partner ACS 8.0 On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123 linuxguy...@gmail.com wrote: I'm looking for ideas for building a innovative, powerful home phone system. Something that does voicemail well, integrates cell phones into

[asterisk-users] Linksys/Cisco 504G randomly restarts

I have 3 Linksys/Cisco 504G phones they keep restarting at what seems to be random. Sometimes as short as 6 minutes. FW version is 7.4.3a I have searched and tried disabling FW check and all related settings. I also extended all the default 3600 resync checks to a lot longer. TIA CF --

Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian va...@arminco.com wrote: On 8/15/11 10:46 PM, C F wrote: I have 3 Linksys/Cisco 504G phones they keep restarting at what seems to be random. Sometimes as short as 6 minutes. FW version is 7.4.3a I have searched and tried disabling FW check

Re: [asterisk-users] FAX Issues

On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote: Hi, I would like to make sure I got it right: 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent fax from the PSTN and have anther FAX machine answer to it even if it is behind asterisk. This works

Re: [asterisk-users] FAX Issues

On Tue, Aug 9, 2011 at 6:00 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Tue, Aug 9, 2011 at 5:21 PM, C F shma...@gmail.com wrote: On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote: Hi, I would like to make sure I got it right: 1. Asterisk 1.4 doesn't support FAX

Re: [asterisk-users] FAX Issues

Guys in my opinion this thread has been very productive. Which proves one thing, as many people you are going to ask about faxing with asterisk that many opinions you are going to get (maybe even add +1 opinions :P). In the end it depends on your experience, hence I asked the OP to try for

Re: [asterisk-users] sip attacks

How long ago was the last block from fail2ban? What could be is that the attacker hasn't yet realized that he has been blocked and is still trying, which although blocked by iptables it is still coming down the line for attempted connections. On Sun, Jul 31, 2011 at 7:04 PM, Dave George

Re: [asterisk-users] Security questions

It's not bad but it wont prevent flooding your box with register attempts and spoofing a user agent is trivia at best. On Sat, Jul 23, 2011 at 9:09 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hello everybody!   I'd like to heard from those with more experience in Security if the

Re: [asterisk-users] Securing Asterisk

On Sat, Jul 23, 2011 at 1:38 PM, CDR vene...@gmail.com wrote: I beg to differ. Digium is hiding from the real world and somebody is Because you have no clue how to secure a box its someone elses fault? going take the software and run with it. My customers lost in excess of $50.000 and cut my

Re: [asterisk-users] 10.0.0 better than 2.0.0?

Since this change I started measuring temperature in Rankine. Its now 592.67 degrees here (south NJ). On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield t...@mountifield.org wrote: I read Kevin's piece in asterisk-announce about the new numbering scheme, and saw in svn-commits some tagging of

Re: [asterisk-users] FAX with SIP

Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu

Re: [asterisk-users] REALY strange issue with making calls biside 2 phones

what does sip show peers say? On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons mat...@jekabsons.lv wrote: Thats my issue, i hope someone could suggest something: Phone A - Phone B == Using SIP RTP CoS mark 5    -- Executing [01@default:1] Dial(SIP/00-0076, SIP/01) in

Re: [asterisk-users] Conference feature

Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and

Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE

Tzafrir, Whats up with this 1.2 vs 1.8 signature? On Thu, Jun 16, 2011 at 3:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Hi, I hope this is not rude of my part. I normally avoid answering mails that relate in such way commercially to hardware. This list is non-commercial If you

Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!

You know I'm a flight engineer but non of the airlines wanted to hire me because other than the self proclaimed title I have no clue how to operate or maintain an aircraft. The dictionary is probably wrong you should patch libpri On Tue, Jun 14, 2011 at 11:43 AM, bilal ghayyad bilmar...@yahoo.com

Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!

Its probably not a bug so don't apply this patch. No D-Channel means it cant sync up. It could be related to anything but the least likely is that its a bug in libpri or dahdi. Just go thru your configs, check and double check the cabling. On Tue, Jun 14, 2011 at 7:27 PM, bilal ghayyad

Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files

You gotto love this this guy. You can almost predict what his next question is. On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I need to create the needed files for the Cisco Phones to be placed in the TFTP server to be able to  register on Asterisk. I

Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files

-1892 Web: http://www.evaristesys.com/ On Jun 12, 2011, at 5:59 PM, C F shma...@gmail.com wrote: You gotto love this this guy. You can almost predict what his next question is. On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I need to create the needed

Re: [asterisk-users] call files .vbs

I'm the original author of said VB Script. Steve is right, I had lots of errors - related to the fact that asterisk watches it too closely and reads the files even before they are complete - and have since updated it that it first dumps it to a temp directory, then use a bash script on the linux

Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

+1 /@ \ \ ___ \ (__O) \ (@) \ (@) \ (__o)_\ \\ On Tue, May 10, 2011 at 4:23 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: I'll keep this brief because I don't want to come across like any more of an a$$ than I absolutely have to, especially

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 3, 2011 at 1:09 AM, A E [Gmail] all.efor...@gmail.com wrote: On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote: Just from my experience with different DBs, stay away from BLOB data types as much as possible. Hi CF, any particular reason why? I've had a good experience

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

On Mon, May 2, 2011 at 11:46 PM, || dave cantera Mobile david.cant...@ibsonecall.com wrote: I've been away from asterisk for a while since 1.4.16 and only installed 1.6 once to run a test... can someone recommend what the best version to install is and the recommended CPU/motherboard for an *

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

Just from my experience with different DBs, stay away from BLOB data types as much as possible. On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if

Re: [asterisk-users] PAP2T auto answer?

The answer function on an analog line is accomplished by going off hook. Unless the line is controlled by an automated device (like answering machine) someone has to physically take the device off hook to answer it. The ATA has no way to do it as all it gives is the FXS signalling. What exactly

Re: [asterisk-users] Asterisk as a Condo door opener/intercom

that would have the features you suggest for $400. Doesnt exist on planet earth not even with Asterisk. The closest you'll get to your mentioned price is a commercial solution. --Don On Behalf Of C F Sent: Monday, April 11, 2011 11:43 AM Search the lists. Some hints: Viking electronics makes

Re: [asterisk-users] Asterisk as a Condo door opener/intercom

Search the lists. Some hints: Viking electronics makes a door box that connects to any analog line (IIRC e-20). They also make a DTMF keypad that integrates in series with any analog line. They might also make a door box with a DTMF keypad on it. Sandman makes a relay that will get energized when

Re: [asterisk-users] Removing Polycom Transfer Softkey

. This sounds like it turns on and turns off the call forwarding feature on the phone.  I can try it out Monday, but I don't see where it has any relation to transfer (both attended and blind). On 03/27/2011 08:43 PM, C F wrote: In phone.cfg set the following line to divert.fwd.1.enabled=0 from

Re: [asterisk-users] Removing Polycom Transfer Softkey

Look at page 311 in that manual If you disable the soft keys and then reassign the hard key it should - at least in theory - be possible to accomplish. On Tue, Mar 29, 2011 at 11:46 AM, C F shma...@gmail.com wrote: Sorry, for some reason I misread it as the forward feature. On Sun, Mar 27

Re: [asterisk-users] Removing Polycom Transfer Softkey

In phone.cfg set the following line to divert.fwd.1.enabled=0 from: divert.fwd.1.enabled=1 For more info check page 323: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf On Fri, Mar 25,

Re: [asterisk-users] What is the most stable version of asterisk?

- From: C F [mailto:shma...@gmail.com] Sent: Thursday, March 24, 2011 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? I use mainly 1.2 with great success, mostly restarts are due to power outages. I

Re: [asterisk-users] What is the most stable version of asterisk?

I use mainly 1.2 with great success, mostly restarts are due to power outages. I recently started to upgrade to 1.4, so far so good. Too early to say, the longest running 1.4 is only 234 days. While I have had 900+ days with 1.2 --

Re: [asterisk-users] Asterisk

I like the subject. On Sun, Mar 20, 2011 at 4:03 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have couple of questions regarding Asterisk. a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone numbers? b) Does it Support VoiceXML ? c) What PRI Card is

Re: [asterisk-users] Metaswitch to Asterisk problems

Make sure you get some DNIS, on the meta you might have to play around to get it, it might come in in the form of some sip headers or as asterisk expects it as an extension. In any event, once asterisk knows which extension (DID) it belongs to just send it to VM. On Thu, Mar 10, 2011 at 10:02

Re: [asterisk-users] How do I find a phone numbers issued by Rogers?

Call them. On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn robert.augus...@linqone.com wrote: Hi, Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? Thanks, Sincerely, Robert Augustyn --

Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

, Shaun Ruffell sruff...@digium.com wrote: On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote: On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote: On 2/21/11 4:46 PM, C F wrote: I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work

Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

This worked. Thank you all for your help. On Wed, Feb 23, 2011 at 1:42 PM, Greg Woods g...@gregandeva.net wrote: On Wed, 2011-02-23 at 09:56 -0500, C F wrote: This is the closest thing I was able to find in my wctdm.c file:         if ((blah 0xf) == 2) {                 /* ProSLIC 3215

Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

How/Where would I do that? TIA CF On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote: On 2/21/11 4:46 PM, C F wrote: I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: Zapata

[asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

this have to do with the fact that the module is way newer than the card? TIA C F -- _ -- 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] Hide the plain text password

On Tue, Feb 15, 2011 at 10:31 AM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Tuesday, February 15, 2011 9:29 AM To: Asterisk Users Mailing List - Non

Re: [asterisk-users] Hide the plain text password

Security through obscurity does not work with open source software. What a bold statement, are you telling me it works with closed source software? :P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Transfer Device Data

${BLINDTRANSFER} should hold the device name of the one doing the blind transfer. On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock murdo...@gmail.com wrote: Hello! I am trying to find out the device name and/or other identifying data to be used in a context when a device transfers the call to

Re: [asterisk-users] PRI voice optimization

On Fri, Feb 4, 2011 at 12:41 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from

Re: [asterisk-users] end a call after a specific time period

Channel              Location             State   Application(Data) SIP/NTT00-   99449046902115@vicid Down    AppDial((Outgoing Line)) Local/99449046902115 99449046902115@defau Up       Dial(SIP/NTT00/449046902115||o Local/99449046902115 8302@default:2       Up      Playback(conf)

Re: [asterisk-users] Unknow T callerid

I guess that was the CallerID transmitted by the calling channel. On Mon, Jan 24, 2011 at 7:31 PM, Jose Flores Galicia floj...@gmail.com wrote: Hi List. Have any of you guys ever see an incoming call throught Dahdi channel which has an callerid T. I know whenever is a private call, it shows

Re: [asterisk-users] end a call after a specific time period

I believe absolute timeout will do that. http://www.voip-info.org/wiki/view/Asterisk+func+timeout On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello all, I am trying to end a call after a specific time period for that reason i have tried various options

Re: [asterisk-users] Top Posting

On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com 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. Its not lazy at all its cutting through the fluff and finding the emails worth while.  

Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a

Re: [asterisk-users] (no subject)

Anyone going to remove this spammer/scammer? 2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com: http://www.barenakedbabies.com/shop/images/images.html -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Voicemail Forwarding

Is that user trying to forward to xxx in the same context? On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in

Re: [asterisk-users] How to find , internal, external inbound or outbound

This list is being attacked by some Khaled I guess. How can we stop him? On Fri, Dec 17, 2010 at 9:34 AM, Khaled W. Chehab kche...@xplorium.com wrote: Hi, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP

Re: [asterisk-users] Version compatibility question...

Thanks again Kevin Have a wonderful day :) On Tue, Dec 7, 2010 at 8:01 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/06/2010 08:12 PM, C F wrote: Thanks Kevin. Upto which version fo Dahdi works with 1.4.x? If I understand your question properly, all versions of DAHDI are compatible

Re: [asterisk-users] Version compatibility question...

Thanks Kevin. Upto which version fo Dahdi works with 1.4.x? On Mon, Dec 6, 2010 at 6:25 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/05/2010 08:25 AM, C F wrote: Is there any version matching doc? since it was changed to Dahdi I don't really know which version works with which

Re: [asterisk-users] Version compatibility question...

Is there any version matching doc? since it was changed to Dahdi I don't really know which version works with which. On Sun, Dec 5, 2010 at 12:35 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Dec 02, 2010 at 09:09:25PM -0300, equis software wrote: Hi, Could I install Asterisk

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