Re: [asterisk-users] Asterisk 1.6 dahdi only?

> Wilton Helm wrote: > > [snip] >> >> My conclusion after installing a worthless * demo (that actually does >> allow two SIPs to talk to each other) is that Asterisk is not of any >> value to anyone other than a person who makes a full time career out >> of running Asterisk systems. I've installe

Re: [asterisk-users] USA BRI -- any hope at all?

I'm actually having trouble understanding what people would order BRI for over POTS lines. The only thing I ever used ISDN for was net access, and it was trumped by DSL a decade ago. Do you get some extra service with your 2B service over ordering two POTS lines? j On Wed, 28 Jan 2009, Fran

Re: [asterisk-users] Looking for SIP loud ringer

Here is the bomb: http://www.clarityproducts.com/products/listing/item3200.asp 95Db :) Plug this into a cheap ATA as was suggested earlier. Solution should be about $100. j On Wed, 28 Jan 2009, Brent Vrieze wrote: > If you know anyone with electronic experience you could take the speaker >

Re: [asterisk-users] ATA recommendation - was: Looking for SIP loud ringer

On Wed, 28 Jan 2009, Mike wrote: > My previous question brings me to this: > > > > I know there are plenty of SIP ATA, but is there one that is particularly > recommended that answers (as many of) the following needs: > > > > 1) As cheap as possible > > 2) Allows for auto-provisioning/configurati

Re: [asterisk-users] USA BRI -- any hope at all?

On Wed, 28 Jan 2009, Karl Fife wrote: > One problem with BRI adoption has no doubt been the need for external power > to the NT1 or TA. > > Obviously analog loops are powered by the CO, so much of the benefit of > ISDN-BRI as the first voice circuit is eroded away for a large percentage of > the

Re: [asterisk-users] Callback / Camp / Extention Free notify?

On Thu, 29 Jan 2009, Daniel Johnson wrote: > pdha...@optusnet.com.au wrote: >> Funnily enough, most people install phones with BLF lamps, on install >> something like hudlite/FOP/etc so you know if the person is on the phone >> before you call them.. >> >> PaulH > > Hi Paul, > > Yes I have see

Re: [asterisk-users] 32 bit server is ok?

Nope - 286 didn't have protected memory access mode, which is key to *nix kernels. j On Thu, 29 Jan 2009, Danny Nicholas wrote: > If you're not GUI-ing, you could theoretically run * on a 286 since Linux > doesn't have the overhead of Windows. > > > > _ > > From: asterisk-users-boun...@li

Re: [asterisk-users] 32 bit server is ok?

This thread made me nostalgic - see this: http://en.wikipedia.org/wiki/MINIX I took a course based on MINIX (as did Linus Torvald) back in 1989 and recall building symbolic links into its kernel as part of a class project. On a 386SX I built in my dorm room. j On Fri, 30 Jan 2009, Jeff

[asterisk-users] TAPI and Asterisk

Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound

Re: [asterisk-users] TAPI and Asterisk

/thirdlane-dialer This seems to be just the ticket... Cheers, j On Fri, 30 Jan 2009, Jeff LaCoursiere wrote: > > Funny how a topic will come up that you have never dealt with before, and > suddenly it comes up from multiple directions at the same time. I was > recently involved in a m

Re: [asterisk-users] Ideas on how to convert spoken name to text (orwav to text)..speech recognition software?

This would work if you only care that you get very rough phonetic spellings as Don implied. If you think about it humans cannot do any better. I know personally - I have to spell my name all the time. Perhaps your app could ask them to spell their name, which actually has a shot at reliabil

Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine

I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with tw

Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine

> on that IP, it will give me all the schema related to this account. Sometimes > I need to use another schema for some calls, I am not able until send for the > provider from another IP. > > Did u get what I need? > Regards > Bilal > > > --- On Sun, 2/1/09,

Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine

indaddr? The problem, > for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. > > Mike > >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >> boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere &g

Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine

peer. >> >> Mike >> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >>> boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere >>> Sent: Sunday, February 01, 2009 14:56 >>> To: bilal ghayya

[asterisk-users] RBS T1 DID issue

Howdy, New installation, trying to connect an RBS T1 with AMI/D4 framing and E&M Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox 2.6.2.1). Outbound calls work fine, but inbound calls fail to read the DID information, and with debug set to 10 I get the following: [Feb

Re: [asterisk-users] RBS T1 DID issue

em_w channel => 1-24 Thanks, j On Mon, 2 Feb 2009, Barton Fisher wrote: > you need to port you zaptel.conf & zapata.conf (might be > channel-additional.conf in trixbox) > > Bart > > - Original Message ----- > From: "Jeff LaCoursiere" > To: > S

[asterisk-users] n-way conferencing

Can anyone suggest a SIP phone that allows conferencing of more than 3 parties locally (i.e. not as part of a conference bridge)? I didn't realize Polycoms are limited to three. TIA, j ___ -- Bandwidth and Colocation Provided by http://www.api-digi

Re: [asterisk-users] n-way conferencing

; d...@cognation.net > +1-212-203-4357 New York > +61-2-9016-5642 (Sydney in-dial). > +44-20-3129-6001 (London in-dial). > > >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >> boun...@lists.digium.com] O

Re: [asterisk-users] RBS T1 DID issue

switched it to "Digitone", which seems to have solved my issue collecting DID digits (I am using em_w signalling in zapata.conf to match Digitone - anyone know if that is the "most" correct?) Thanks, j On Tue, 3 Feb 2009, Jeff LaCoursiere wrote: > > Howdy, > >

Re: [asterisk-users] Autodialler query

Sounds scammy. Do the "customers" know that this autodialer will be charging them? j On Thu, 5 Feb 2009, Kinjal Dixit wrote: > Sriram: > > whats going on here?? > > unless you are developing a vas, in which case, the provider for whom you > are doing this will have to help you. each provider

[asterisk-users] Credit Card processing machines

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card.

Re: [asterisk-users] Credit Card processing machines

A bit of hopefully happy news - the Linksys 2102 has a feature called "modem pass through mode" which can be accessed by prepending *99 to the call. Anyone ever used this? Sounds like that might help with faxing as well... j On Fri, 6 Feb 2009, Jeff LaCoursiere wrote: > >

[asterisk-users] reinvite

I've never used "reinvite" in systems I have installed to date, and I have finally run across a situation where it would be preferred. A remote office has a flaky Internet connection. With G729 encoding the calls to the central office over the 'net are tolerable. One Linksys 2102 drives two

[asterisk-users] odd disconnects with major company's voice recog

nt voice on the "cannot be completed" message, but the effect of course is the same! Cheers, Jeff LaCoursiere SunFone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us f

Re: [asterisk-users] Dahdi for meetme on AMD64 arch?

On Fri, 2012-01-20 at 15:31 +0100, Johan Wilfer wrote: > > For the next server I will research will further, and test out lxc to > see if it can replace openvz. I found some posts to this mailinglist > about lxc that seems to indicate that it works well, and the methods > are quite similar to ope

Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

On Thu, 2012-01-26 at 10:48 -0600, Tim Nelson wrote: > - Original Message - > > On 01/26/2012 09:46 AM, Tim Nelson wrote: > > > Greetings- > > > > > > I currently have a customer that *requires* key-system functionality > > > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the

Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

On Thu, 2012-01-26 at 18:21 +0100, Patrick Lists wrote: > On 26-01-12 18:08, Jeff LaCoursiere wrote: > [snip] > > > > I'm also very interested in working examples, especially if someone has > > set it up for SIP termination "trunks" rather than Dahdi. > &

Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

On Tue, 2012-01-31 at 14:13 +0100, Gilles wrote: > On Tue, 31 Jan 2012 07:57:22 -0500, "bakko" > wrote: > >yeallink T26 and T28 support OpenVPN too > > Thanks for the infos. > > If someone tried the Snom, Grandstream, or Yeallink, how good is their > OpenVPN connection? > > Using Yealink T-28

Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

On Tue, 2012-01-31 at 11:29 -0500, Doug Lytle wrote: > Jeff LaCoursiere wrote: > > Bummed that it seems to only support one tunnel, though > > > As in you can't register the phone to more then 1 remote Asterisk server > via 2 different VPN tunnels or you can't hav

Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

On Tue, 2012-01-31 at 17:23 +0100, Gilles wrote: > On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere > wrote: > >Using Yealink T-28 with OpenVPN works fine - about three weeks now with > >no issues. Bummed that it seems to only support one tunnel, though. I > >ask

Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

On Thu, 2 Feb 2012, Tzafrir Cohen wrote: Oh, and for the record, you can tunnel practically on top of anything. Just in case you're not familiar with it: IP over DNS (which means you don't even need direct access, and can use proxied DNS queries). http://code.kryo.se/iodine/ I figure you won't

[asterisk-users] OpenVPN design w/ Yealink

d calls to the datacenter that is "up". But that smacks of another single point of failure. Any advice? Thanks! Jeff LaCoursiere SunFone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Vitelity Setup

Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alex

Re: [asterisk-users] Vitelity Setup

On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: > On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere wrote: > > > > Word of warning - I have had a lot of issues with Vitelity's routing. > > Lots of troubles to the Caribbean, lots of troubles with ordinary

Re: [asterisk-users] Vitelity Setup

On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: > On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere wrote: > > On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: > >> On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere > >> wrote: > >> >

Re: [asterisk-users] Digium IP Phones - Teleworker Capability?

On Thu, 2012-06-14 at 16:23 -0600, asterisk users wrote: > > This is pretty good news, overall. To comment on Kevin's points: > > - The end-to-end encryption is important to us, because > client-ID-sensitive information is part of our environment. Something > like built-in OpenVPN would work fo

Re: [asterisk-users] Digium IP Phones D40

On Mon, 2012-06-25 at 10:45 -0500, Kevin P. Fleming wrote: > On 06/22/2012 05:12 PM, bilal ghayyad wrote: > > One of the problems I faced with Polycom is the voice volume and ring > > volume, it is low. > > > > When it rings, even if it is maximum volume, still it is weak. > > When I talk and I se

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

On Thu, 2012-07-12 at 15:49 +1200, Alec Davis wrote: > I've seen similar. > > We tried 4 interfaces. On 4 lans, are these considered to be overlapping? > 192.168.1.1 > 192.168.2.1 > 192.168.3.1 > 192.168.4.1 > Depends on the netmask you use :) Assuming you used /24, so "no", they don't overlap.

Re: [asterisk-users] Grandstream VoIP phones

On Fri, 2012-08-17 at 09:30 -0700, Carlos Alvarez wrote: > On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson > wrote: > My primary interest is security. Grandstream claims their > intermediate and higher-end models support TLS and SRTP. I am > really tired of trying

[asterisk-users] FAX detection in chan_dahdi 1.8.15

Hi, I recently replaced a site that was using 1.4.[mumble] with hylafax/iaxmodem. They have an RBS T1 and were using about half of their 50 DID numbers for "fax to email". This all broke with the new system :( The original chan_dahdi.conf had no mention of "faxdetect", so I assume it was

[asterisk-users] FAX detection in chan_dahdi 1.8.15

Hi, I recently replaced a site that was using 1.4.[mumble] with hylafax/iaxmodem. They have an RBS T1 and were using about half of their 50 DID numbers for "fax to email". This all broke with the new system :( The original chan_dahdi.conf had no mention of "faxdetect", so I assume it was

Re: [asterisk-users] FAX detection in chan_dahdi 1.8.15

> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff > LaCoursiere > Sent: Tuesday, August 28, 2012 3:24 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] FAX de

[asterisk-users] codec priorities

Hello, I am about to start playing with wideband codecs in our lab, and was hoping to get some clarification on a few things. To date I've pretty much forced the use of G.711 on all legs of all calls, and life has been grand. Now we are distributing phones with G.722 and speex capability, and I

Re: [asterisk-users] FAX detection in chan_dahdi 1.8.15

On 09/13/2012 03:20 AM, Olivier wrote: 2012/8/31 Jeff LaCoursiere mailto:j...@sunfone.com>> > -Original Message- > From: asterisk-users-boun...@lists.digium.com <mailto:asterisk-users-boun...@lists.digium.com> > [mailto:asterisk-users-bou

[asterisk-users] T.38 gateway ATA

Hoping for some clarification. I would like to setup a NORMAL (not T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a provider. Some amount of googling today seems to imply that most ATAs are just T.3

Re: [asterisk-users] T.38 gateway ATA

p and control of the endpoints (gateways and ATAs) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: "Jeff LaCoursiere" *Sent*: Monday, September 24, 2012 9:20 PM *To*: "Asterisk User

Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 03:54 AM, Sylvain Rochet wrote: Hi Jeff, On Mon, Sep 24, 2012 at 08:18:53PM -0500, Jeff LaCoursiere wrote: Hoping for some clarification. I would like to setup a NORMAL (not T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a remote asterisk (1.8) server, which

Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:29 AM, Bryant Zimmerman wrote: *From*: "Jeff LaCoursiere" *Sent*: Tuesday, September 25, 2012 11:05 AM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] T.38 gateway ATA On

Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:34 AM, Sylvain Rochet wrote: Hi Jeff, On Tue, Sep 25, 2012 at 10:11:27AM -0500, Jeff LaCoursiere wrote: Aha! That is what I am looking for. What firmware rev are you using on the 3102? 5.1.12 on SPA8000, 5.1.10(GW) on SPA3102, there is probably new releases from Cisco, at

Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:44 AM, Danny Nicholas wrote: I would recommend the OBI110. -- _ -- 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] T.38 gateway ATA

On 09/25/2012 03:02 PM, Bryant Zimmerman wrote: Jeff The grandstream units support both Relay and passthru depending on how you set them up. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth a

Re: [asterisk-users] Tips for installing and configuring Digum cards

On 10/11/2012 09:34 AM, Mitch Claborn wrote: In case the moderator doesn't approve my post with the attachment, below is a quick and dirty transcription of the order form. Customer connecting equipment: CSU/DSU Circuit: DS1 Line coding: B8ZS Framing: ESF Jack type: RJ48X / Smartjack (will that

Re: [asterisk-users] high capacity analog <-> sip gateway

Agree with 24 port being the max for a single device. In that vein I just deployed a handful of Grandstream 24 port FXS devices that seem to be working well at a decent price point. I don't normally recommend Grandstream for anything, and in the past we have only deployed Audiocodes for thi

[asterisk-users] USB FXS device

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 spendin

Re: [asterisk-users] USB FXS device

On 10/31/2012 01:38 PM, 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

Re: [asterisk-users] USB FXS device

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

Re: [asterisk-users] USB FXS device

On 10/31/2012 02:00 PM, jon pounder wrote: 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

Re: [asterisk-users] USB FXS device

On 10/31/2012 04:43 PM, Benny Amorsen wrote: Jeff LaCoursiere writes: The basic question was "has anyone made a USB FXS device work with asterisk". Now that I have additionally defended my architecture decisions, can anyone actually answer the question? The Open USB FXS project

Re: [asterisk-users] USB FXS device

On 11/01/2012 01:00 PM, Mahendra Dobariya wrote: i manage to make call and receive call , IVR , GSM gateway less than 100 $. https://www.facebook.com/photo.php?fbid=451312488254098&set=a.435627683155912.115286.11260518622&type=3&theater see the pic, and let me know if it is useful to you.

Re: [asterisk-users] USB FXS device

On 11/01/2012 04:37 PM, Benny Amorsen wrote: Jeff LaCoursiere writes: Nifty! Love this Raspberry Pi. I keep thinking of new things I want to do with it. If I could only clone myself. I have a "video doorbell" project at the top of the list, if I don't find a USB FXS device :)

Re: [asterisk-users] USB FXS device

On 11/04/2012 04:17 AM, Andreas Sikkema wrote: Draytek Vigor2110Vn Sadly this doesn't seem to do OpenVPN, though it does several other flavors we might be able to support. Thanks for the tip! Will be looking into it. Cheers, j -- _

Re: [asterisk-users] Managing complex setups with Asterisk

On 11/07/2012 01:06 PM, Joshua Colp wrote: martin f krafft wrote: also sprach Joshua Colp [2012.11.07.1831 +0100]: Peer names have to be distinct, this is just a fundamental design element of chan_sip. What a lot of people end up doing is instead of treating peers as people they treat them as

Re: [asterisk-users] Managing complex setups with Asterisk

On 11/07/2012 02:16 PM, Johan Wilfer wrote: 2012-11-07 20:49, Jeff LaCoursiere skrev: Just to chime in, if you REALLY want multi-tenant, it is super easy and surprisingly efficient to use kernel level virtualization to run multiple instances of asterisk (and even FreePBX). We use LXC to do

Re: [asterisk-users] Managing complex setups with Asterisk

On 11/07/2012 05:20 PM, Jeff LaCoursiere wrote: On 11/07/2012 02:16 PM, Johan Wilfer wrote: 2012-11-07 20:49, Jeff LaCoursiere skrev: Just to chime in, if you REALLY want multi-tenant, it is super easy and surprisingly efficient to use kernel level virtualization to run multiple instances of

Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

You can also (in mysql) tag a column as "unique": alter table blah add unique(column_name); This doesn't add a key AFAIK unless you ask it to be a key: alter table blah add unique key(column_name); If you will never refer to a row using the auto-increment column, why have it? No problem ha

Re: [asterisk-users] Users list email totals by year .

On 12/29/2012 05:20 PM, Mr. James W. Laferriere wrote: 2003, 24471 2004, 48608 2005, 59116 2006, 41215 2007, 26414 2008, 20746 2009, 18304 2010, 14948 2011, 11588 2012, 7542 If you remove the top-posting thread, it may cut it in half again. j -- _

Re: [asterisk-users] Verizon SIP "trunking" Field Trial

On 01/03/2013 09:56 AM, Carlos Alvarez wrote: On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young > wrote: Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same

[asterisk-users] DID providers

Howdy, Looking to port numbers that we currently own in the US Virgin Islands to *any* carrier that can do it in the states. XO says it is out of their scope. Our normal DID carrier (IP Comms) apparently uses XO... Looking for recommendations? Thanks, j -- __

Re: [asterisk-users] Asterisk - Soundcard - Recording?

I'll take a stab, since you said no GUI and also USB based mic. Raspberry Pi project? I'm interested in this vein as well, especially after the recent post about voice recognition. I was thinking that Raspberry Pi's with mics could live around my house and all have dedicated always-open cha

[asterisk-users] voice recognition voicemail to email

I fuzzily recall someone posting a script that shuffled off voicemails to Google for conversion to text that could then be emailed. Anyone have any luck with that? Anything new out there? j -- _ -- Bandwidth and Colocation

[asterisk-users] T.38 termination

using Hylafax/t38modem with a US based termination provider? I don't care about inbound at the moment - just the ability to use "sendfax" to shoot out documents on demand. Thanks, Jeff LaCoursiere StratusTalk, Inc. -- __

[asterisk-users] Polycom BLF

ot;lose" its hint for a phone once in a while, and report it as unreachable, even though it can easily make and receive calls from it. Am I tilting at windmills? Is this really unstable or has someone made it work so

Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension

On Wed, 15 Jun 2011, Florent THOMAS wrote: Do you know some devices that aren't so "locked"? None of them are locked by default - it's only the service providers that lock them into their own networks - so if you buy anything from an online supplier that doesn't co

Re: [asterisk-users] TDM400p susceptible to EMI?

On Wed, 2011-07-13 at 17:19 -0400, Andrew Latham wrote: > On Wed, Jul 13, 2011 at 5:16 PM, Tim Nelson wrote: > > - Original Message - > >> > On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards > >> > wrote: > >> >> I have a TDM400p with 3 fxs and 1 fxo daughter cards. > >> >> > >> >> It's in a

Re: [asterisk-users] pickupgroup

On Thu, 2011-08-04 at 11:20 -0400, Dan Journo wrote: > Hi, > Using 1.4, I see that pickupgroup can only be between 1 and 63. > We run a hosted PBX service and need to give our client access to the > call pickup feature. > I thought that I could simply use the client's ID number for the > pickupgr

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

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 sho

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

On Fri, 2011-08-26 at 12:37 -0600, linux guy wrote: > > Do any of the DECT systems handle multiple incoming phone lines ? > > How do the DECT systems integrate with the voice mail services on an > Asterisk system ? The single line Panasonic that I use doesn't handle multiple phone lines itself,

Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?

On Fri, 2011-08-26 at 13:18 -0600, linux guy wrote: > On Fri, Aug 26, 2011 at 12:55 PM, Ian Pilcher > wrote: > On 08/26/2011 01:02 PM, linux guy wrote: > > I'm looking for 4 to 6 good, inexpensive VOIP handsets for > my home > > asterisk system. > >

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

On Sat, 2011-08-27 at 09:31 +0100, Alan Lord (News) wrote: > On 26/08/11 12:28, linux guy wrote: > > > > Great discussion, all of it. Thanks, people. > > > > How much power does the home asterisk box need ? > > Not much :-) > > I've been running our phone system and home media/storage network on

Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?

On Thu, 1 Sep 2011, RSCL Mumbai wrote: Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP

Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?

On Thu, 1 Sep 2011, RSCL Mumbai wrote: I tried and failed with VirtualBox too.  Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance).  I don't think VirtualBox is up to real-time stuff. We use LXC now,

Re: [asterisk-users] Asterisk on Android?

On Thu, 8 Sep 2011, Cobra 2 wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine.  Chrooted? I don't think anyone was claiming it wouldn't run - just that it wouldn't be useful if it cannot use the phone's radio as a ch

Re: [asterisk-users] Fax from FXS to PRI

On Tue, 2011-09-20 at 20:57 -0500, Don Kelly wrote: > On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett > wrote: > > If I have a 4 port Digium FXS card and a single port PRI card on the > same asterisk box, is it expected that I'd be able to plug a fax > machine into the analog FXS port and have no p

Re: [asterisk-users] No Bull Service Providers

On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing aga

Re: [asterisk-users] No Bull Service Providers

hould have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere wrote:

[asterisk-users] DIDs in Singapore

Can anyone suggest an ITSP with Singapore DIDs and local Singapore termination? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Becoming a CLEC

On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: > The ride is over before it even began A local ILEC here in Canada, > is already offering > Unlimited World service. And this on a Tier 1 network, not the crap > we're use to doing > business on. Choose a different angle before you get anym

[asterisk-users] Errors on RBS T1

local/bin# dahdi_hardware pci::01:03.0 wct4xxp+ 10ee:0314 Wildcard TE410P/TE405P (1st Gen) root@vigw3:/usr/local/bin# ls -ld /usr/src/dahdi* drwxrwxr-x 5 root root 4096 2010-05-17 15:49 /usr/src/dahdi-linux-complete-2.3.0.1+2.3.0 root@vigw3:/usr/local/bin# asterisk -V Asterisk 1.

Re: [asterisk-users] Errors on RBS T1

On Fri, 2011-12-16 at 11:12 -0600, Russ Meyerriecks wrote: > - Original Message - > > From: "Jeff LaCoursiere" > > I am seeing errors accumulate on an RBS T1 and am wondering what to > > make > > of them: > > Could we take a peek at /etc/dahdi/s

[asterisk-users] Diagnosing call hangups

her "Dial" or "Answer" that would allow me to log which side hungup without having to enable debug and plow through a gazillion log lines to find the answer? How do others debug random hangups for customers? Thanks! Jeff LaCoursiere SunFone --

Re: [asterisk-users] Client - registers but unreachable

On Wed, 2011-12-28 at 23:16 -0500, Michelle Dupuis wrote: > I have a softphone I'm trying on a blackberry, that registers on my > Asterisk, can make outgoing calls, but can't receive calls. > > There is very little traffic with this phone (see debug below - as the > phone registers), and sip sho

Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)

On Tue, 4 Jan 2011, Andy Graybeal wrote: The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone

Re: [asterisk-users] Asterisk replying to wrong port for NOTIFY messages

On Wed, 5 Jan 2011, James Lamanna wrote: See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Hi James, I'm sure it would be the NAT translated port on the public side of the customer's firewall... j Thanks. -- James <--

Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver

On Tue, 18 Jan 2011, Asterisk Security Team wrote: Asterisk Project Security Advisory - AST-2011-001 ProductAsterisk SummaryStack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility

[asterisk-users] uptime

Now this is what I call uptime... minipbx*CLI> show uptime System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds Last reload: 8 hours, 3 minutes, 51 seconds Bizarre bug? root@minipbx:~# asterisk -V Asterisk 1.4.37 root@minipbx:~# uname -a Linux minipbx 2.6.32-dockstar #2 Th

Re: [asterisk-users] uptime

On Tue, 15 Feb 2011, A J Stiles wrote: On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote: Now this is what I call uptime... minipbx*CLI> show uptime System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds Last reload: 8 hours, 3 minutes, 51 seconds Bizarre bug?

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

On Wed, 23 Feb 2011, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.

Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

On Thu, 10 Mar 2011, Vladimir Mikhelson wrote: Pay attention, you have permit=172.16.16.0/24 whereas suggestion was permit=0.0.0.0/0.0.0.0 Pay attention? Maybe you should. He is clearly trying to restrict access to the local network, not open it up to the world. j On 3/10/2011 1:48

Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

Thanks. But Like I said,  that's all done. Here's the Endpoint config: [authentication] [basic-options](!)                ; a template         dtmfmode=rfc2833         context=Phones         type=friend         contactdeny=0.0.0.0/0.0.0.0         contactpermit=172.16.16.0/255.255.255.0       

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