Re: [asterisk-users] Xorcom PRI

Turn on PRI debugging and double check your cable. On Mon, Nov 12, 2018 at 3:24 PM Jeff LaCoursiere wrote: > > I've been struggling for a few weeks now with the local telco trying to > bring up a trunk that has been down for a year (hurricanes in the > caribbean). Box is a Dell R710, 16G RAM,

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

Possibly the realm? Thanks, Steve On Sat, Sep 2, 2017 at 3:58 AM, O. Hartmann wrote: > > It might sound stupid and a kind of "noobish", but I have serious trouble > with > registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT > box. > > The

Re: [asterisk-users] Moving call DAHDI from channel X to Y.

I remember seeing something like this a long time ago. If memory serves me correctly it was a problem at the physical layer and a couple of the PRI cables got flipped and plugged into the wrong port. I had to change the configs since I didn't have physical access to the box. Thanks, Steve On

Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello lucab...@lucabert.de wrote: Ashwin Surendran ashwin.surend...@now-health.com schrieb: What settings have you got for directmedia? Could you try nat

Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello lucab...@lucabert.de wrote: Ashwin Surendran ashwin.surend...@now-health.com schrieb: What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no Tried. Peer always unreachable, call not

Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello lucab...@lucabert.de wrote: Zitat von Steve Totaro stot...@totarotechnologies.com: Are you using the wifi on on the cellphone? The peer IP is showing as 192.168.200.3 which is not a routable address. Unless things have changed, double NAT

Re: [asterisk-users] Lost audio on forwarded calls

Asterisk does not need to care. Is it SIP all the way through? Thanks, Steve T On Fri, Oct 3, 2014 at 3:12 PM, Todd R. tjrl...@live.com wrote: OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think

Re: [asterisk-users] PRI timing settings

PRI intense debug should show all you need to fix this. On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net wrote: Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j On 08/20/2014 10:13 AM, Don Kelly

Re: [asterisk-users] T1 Card RED ALARM

Remember to always check your cables first. Thanks, Steve T On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote: Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh

Re: [asterisk-users] VPN SIP Phone | PC Traffic

I did this with SNOM phones and a special firmware a while ago. The trick to get the VPN to extend to the PC port is bridge-utils. Worked very well. On Apr 9, 2014 7:40 AM, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our

Re: [asterisk-users] IAX2 Trunk Encryption

Wireshark. On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote: Ok, I think I am 90%+ there. Note: the configuration or status is the same on both sides unless otherwise noted. I am using RSA keys for authentication and the calls are coming through as

Re: [asterisk-users] IAX2 Trunk Encryption

is not. That just leaves the question of what I need to do to get it encrypted.. Thanks. On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro stot...@totarotechnologies.com wrote: Wireshark. On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote: Ok, I think I am 90%+ there. Note

[asterisk-users] (OT) Phones with STP, DHCP, and/or (T)FTP Issues

I remember having to turn off STP or set portfast on some switch ports to some phones due to the boot sequence and timeouts of some phones a long time ago. Does anyone know which phones, if any still suffer from these problems? I am setting up a lab and want to introduce this problem for the

Re: [asterisk-users] Need more meetme users -- hitting some limit

I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You

Re: [asterisk-users] Need more meetme users -- hitting some limit

On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 21 Mar 2014, Steve Totaro wrote: I found below here: http://www.voip-info.org/ wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you

Re: [asterisk-users] Need more meetme users -- hitting some limit

Is there any good documentation on that process? On Fri, Mar 21, 2014 at 3:36 PM, John Novack jnov...@stromberg-carlson.orgwrote: Steve Edwards wrote: On Fri, 21 Mar 2014, Adrian Serafini wrote: Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm a big fan of

Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

Gateway computers rejects calls like this. I was informed that their carrier rejects the calls because they cannot accurately bill. It seems pretty silly with voip and number portability. Thanks, Steve T On Mar 17, 2014 5:19 PM, Eric Wieling ewiel...@nyigc.com wrote: Often it is

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks

Re: [asterisk-users] send the calls from to servers

POTS? On Thu, Dec 19, 2013 at 1:31 PM, Salaheddine Elharit salah.elharit...@gmail.com wrote: i ask about outbound calls not inbound round-robin best regards 2013/12/19 Eric Wieling ewiel...@nyigc.com Inbound call hunting is handled by your carrier, not Asterisk. -Original

Re: [asterisk-users] Asterisk TON number

to change dialplan? ** ** ** ** Br ** So what are you trying to do specifically? Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Asterisk TON number

dailplan and pattern matching. You can probably do all of that from the Trixbox GUI. If you like Trixbox, check out FreePBX since Trixbox is done. Thanks, Steve Totaro On Wed, Sep 25, 2013 at 9:30 AM, Endri Stefani endri.stef...@plus.alwrote: Hi guys ** ** Thanks a lot, I am just

Re: [asterisk-users] Asterisk TON number

So you are using QSIG and connecting your Asterisk box to a legacy PBX over PRI E1? Did you try unknown? Do you need to use QSIG (over euroisdn for instance)? Thanks, Steve Totaro On Wed, Sep 25, 2013 at 10:33 AM, Endri Stefani endri.stef...@plus.alwrote: Hi Steve

Re: [asterisk-users] Asterisk TON number

The call is being placed, is it not? Again, I know you are trying to change the TON but what are you trying to accomplish and what is failing. It seems like you are dialing 1000 and that is being sent on the wire. Thanks, Steve Totaro On Wed, Sep 25, 2013 at 10:37 AM, Steve Totaro stot

Re: [asterisk-users] Sip-Client / type=peer / Why can this client place calls?

On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote: Hi, I am using Asterisk 11.5.1. As far as I understood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls

Re: [asterisk-users] Conference calls wont traverse my trunk

On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote: My conference call wont go thru my SIP trunk. I may be missing a dialplan configuration setting as my PCM phone to phone calls go over the (GSM) tunk. The server with the conference: exten =

Re: [asterisk-users] I need a second opinion on a new phone system deployment

http://red-fone.com http://red-fone.com/products-new/fonebridge/ might be a good place look and see if other ideas pop up. They have good products. I am not affiliated with them, just a happy user on a couple of deployments. On Fri, Jun 14, 2013 at 11:43 AM, Nunya Biznatch

Re: [asterisk-users] asterisk fax in debian

of it? I would go with HylaFAX. FAX is an art with any VoIP solution. The best art I have done and seen turned out to use HylaFAX. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)

On Thu, Jun 13, 2013 at 9:23 AM, James Bensley jwbens...@gmail.com wrote: Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs,

Re: [asterisk-users] A problem with IAX2

similar issue, but it is suspended and, like stated there, the problem is very hard to reproduce. See: https://issues.asterisk.org/jira/browse/ASTERISK-21762 -- משיח NOW! Use SIP and never look back. Thanks, Steve Totaro

Re: [asterisk-users] OC3/STM-1 Line Card

Adtran MX2800 is rock solid. Save some money and use NFAS. Thanks, Steve Totaro On Sun, Jun 9, 2013 at 10:11 PM, Nick Khamis sym...@gmail.com wrote: Thank you so much for your responses!!! With this route we would have to manage so many * boxes with T1s, not to mention, the hit we would

Re: [asterisk-users] Asterisk Calendar integration suggestions

Without knowing requirements, Sugar CRM seems to be the most supported. Thanks, Steve Totaro On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us j...@millican.us wrote: Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender

Re: [asterisk-users] Redirect incoming call to SIP trunk.

be happy to see your way to get out of it I would bet you that is exactly what he did. This list has died off so much because you can find almost every answer in the archives now. Thanks, Steve Totaro -- _ -- Bandwidth

Re: [asterisk-users] Wierd question - Give me your opinion please

On Mon, Feb 4, 2013 at 11:11 PM, Jared Baxley jared.bax...@gmail.com wrote: Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge

Re: [asterisk-users] DIDForSale spam

will refrain myself on any further unproductive communication. Happy new year to you all. On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote: +1 here. On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 7:03 PM, chris tknch

Re: [asterisk-users] DIDForSale spam

So what asterisk issue do you have? Let's fix it. On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler rwhee...@artifact-software.com wrote: That does not solve any asterisk issue that I have. On 10/01/2013 1:32 PM, Carlos Alvarez wrote: On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire

Re: [asterisk-users] DIDForSale spam

A tier one provider. On Thu, Jan 10, 2013 at 3:44 PM, Carlos Alvarez car...@televolve.com wrote: Hopefully it's not, What is the best DID provider for Asterisk... On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro stot...@totarotechnologies.com wrote: So what asterisk issue do you have? Let's

Re: [asterisk-users] Streaming/Recording audio

On Wed, Jan 9, 2013 at 2:54 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, For some reason I did not receive any replies related to my question by mail, but I found the topic back on the online mailing archives. I hope by supplying the same subject this email will be logged in my previously

Re: [asterisk-users] Streaming/Recording audio

On Wed, Jan 9, 2013 at 4:16 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 2:54 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, For some reason I did not receive any replies related to my question by mail, but I found the topic back on the online mailing

Re: [asterisk-users] DIDForSale spam

On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote: On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a

Re: [asterisk-users] new user help required to build voice recorder with asterisk

compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz

Re: [asterisk-users] new user help required to build voice recorder with asterisk

be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com

Re: [asterisk-users] new user help required to build voice recorder with asterisk

Mixmonitor also muxes the two sides of the conversation after hangup. That is quite a bit of I/O for 60 simultaneous calls lasting an average of 5-15mins On Wed, Jan 2, 2013 at 9:59 AM, Steve Totaro stot...@totarotechnologies.com wrote: It depends on what you do with them. Years ago, 60 calls

Re: [asterisk-users] Top Posting

On Wed, Jan 2, 2013 at 11:00 AM, Richard Kenner ken...@gnat.com wrote: I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting

Re: [asterisk-users] Top Posting

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

Re: [asterisk-users] Top Posting

On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 12:20 PM, Steve Totaro wrote: 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

Re: [asterisk-users] Asterisk as answering machine

Problematic at best. Just make a phone an extension and allow that to ring in a hunt group. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 3:30 PM, Ron Wheeler rwhee...@artifact-software.com wrote: I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone

Re: [asterisk-users] Top Posting

On Wed, Jan 2, 2013 at 3:46 PM, jon pounder j...@inline.net wrote: 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

Re: [asterisk-users] Top Posting

Yeah. I never really got the whole fanatical top vs bottom thing. Whatever, I have answered way more than my fair share of free questions (as in beer). The person asking was always quite happy to get a meaningful and helpful reply, no matter where it was in the body of the content. Why people

Re: [asterisk-users] Paging for Praying

On Fri, Dec 28, 2012 at 11:35 PM, John Novack jnov...@stromberg-carlson.org wrote: Shaun Ruffell wrote: On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote: On 12/28/2012 08:13 PM, Steve Edwards wrote: Please don't top-post. If you don't know what that means, please consult

Re: [asterisk-users] CHANNEL(t38passthrough) is 0

Do you have reinvite allowed? That was an issue on one of my installations if I am remembering correctly. Any debug, logs, confs that would help? Thanks, Steve Totaro On Thu, Dec 27, 2012 at 12:15 PM, Eric Wieling ewiel...@nyigc.com wrote: Setting directmedia=no does not help. The calls

Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

On Sun, Dec 9, 2012 at 2:54 PM, Stephen Brown stephen.brow...@gmail.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 So a friend of mine and I setup a static key based point to point OpenVPN connection from my box to his for the express intent of carrying IAX traffic encrypted.

Re: [asterisk-users] PRI can receive calls but cannot dial out

cable to connect cpe to net. Spans should come up and you should be able to simulate the telco and test everything out in both directions. Finally, call Digium and your telco if you are able to do the above with no problems. Thanks, Steve Totaro

Re: [asterisk-users] Hacked by Microsoft?

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

Re: [asterisk-users] Wireshark AMI Dissector

On Fri, Oct 26, 2012 at 2:52 AM, Olle E. Johansson o...@edvina.net wrote: 23 okt 2012 kl. 22:31 skrev Kristian Kielhofner k...@kriskinc.com: Hello everyone, Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector? Decode as telnet and display filter telnet.data kind of

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

/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

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

On Thu, Oct 25, 2012 at 5:01 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 25/10/12 9:49 pm, Justin Killen wrote: What would be the advantage of using 100 single units vs. just buying VoIP phones? That doesn't seem very cost effective to me in the long run. In older buildings with

Re: [asterisk-users] How to tie orders taken to specific CDR records

On Thu, Oct 25, 2012 at 12:18 PM, Mitch Claborn mitch...@claborn.net wrote: Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that

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

. If they have lots of lights and a display, they are most likely digital phones. What kind of PBX and phones do you have. Before digital, phones needed 25 pair to control the phone's various lights, lines, mwi. Thanks, Steve Totaro

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

On Thu, Oct 25, 2012 at 5:09 PM, jon pounder j...@inline.net wrote: 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

Re: [asterisk-users] Fully utilise all PRIs in a DAHDI group

://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F Taken from the wiki searching with the exact terms you used. http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Thanks, Steve Totaro Dialing a GroupIn the Zap Channel

Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

. This doesn't appear to be the problem though. It may be. Did you try saving a change in FreePBX and applying it? It seems more like a FreePBX config error that should be overwritten by FreePBX database to flat files. Thanks, Steve Totaro

Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote: I was following Digium's instructions to the letter to install g729. but upon telling asterisk to load the module, the system hung

Re: [asterisk-users] Odd cracking with SIP-DAHDI

On Tue, Oct 16, 2012 at 1:31 PM, Richard Kenner ken...@gnat.com wrote: We recently set up a SIP trunk between an office in NY running Asterisk and an office in Paris (running Alcatel). All works fine if a SIP phone on the NY system talks to the Paris PBX. But if something on DAHDI (a PRI or

Re: [asterisk-users] Odd cracking with SIP-DAHDI

and possibly ask your question on this site. http://www.alcatelunleashed.com/search.php?keywords=SIP+trunking+with+Asterisk Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] SIP, Polycom, Asterisk - VPN

firmwares. I have even run Asterisk on these little gems. Some SNOM phones have a Linux/OpenVPN firmware and you can actually bridge the WAN/LAN ports and use the phone as a gateway. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Who said asterisk is not to the task

On Sat, Sep 29, 2012 at 6:49 AM, Markus unive...@truemetal.org wrote: Am 29.09.2012 10:49, schrieb resea...@businesstz.com: [tz-ivr01 ~]# uptime 11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57 Sharing is caring Is that a Quad Core CPU in your box? PS: Yes,

Re: [asterisk-users] using analog phones

if you find that they are digital. 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

Re: [asterisk-users] Video conferencing?

On Thu, Jul 26, 2012 at 2:22 PM, Jonathan Rose jr...@digium.com wrote: Ken D'Ambrosio wrote: From: Ken D'Ambrosio k...@jots.org To: asterisk-users@lists.digium.com Sent: Wednesday, July 25, 2012 1:24:50 PM Subject: [asterisk-users] Video conferencing? Hi, all. I'm 99% sure that

Re: [asterisk-users] app_rpt

DAHDI did. I may fire up a Debian Lenny VM and see if the fork with the patches match up and work, and then if app_rpt and app_radio compile or throw an error. The latest all in one ISO uses CentOS 5.7. Thanks, Steve Totaro

Re: [asterisk-users] app_rpt

2012/3/9 Paul Belanger pabelan...@digium.com On 12-03-09 03:18 AM, Márkus Béla wrote: how can I add/enable app_rpt module to Asterisk 1.8? Make sure DAHDI is installed. However, there is a patch on reviewboard[1] that will see this module be removed from asterisk. The code is out-dated

Re: [asterisk-users] app_rpt

On Fri, Mar 9, 2012 at 8:52 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: 2012/3/9 Paul Belanger pabelan...@digium.com On 12-03-09 03:18 AM, Márkus Béla wrote: how can I add/enable app_rpt module to Asterisk 1.8? Make sure DAHDI is installed. However, there is a patch

Re: [asterisk-users] app_rpt

On Fri, Mar 9, 2012 at 4:10 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/09/2012 02:56 PM, Josh Freeman wrote: The most current patched Asterisk, along with the most current app_rpt, can be found at

[asterisk-users] Fwd: Do you know how Asterisk came to be?

Apologies for the top post, something is screwed up with my email client, will fix it soon. What a BS story that I have debunked many times. A used Key System could be purchased for a few hundred dollars, a much better investment then writing your own PBX from scratch. A company that is

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. Thanks, Steve Totaro

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote: On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

one. If you are going to do the job, do it right from the start so that you can grow or change with ease and use real recognized standards. If you are just playing around, do whatever. Actually do whatever, and learn the hard way, I don't care, just trying to help. Thanks, Steve Totaro

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Roger That, I am an IC. I contract with the Government to little ten phone shops. From VA/MD/DC area, I have been contracted and flown in to many large call center locations that were CONUS and OCONUS. My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but my resume speaks

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote: On

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

combined into one stream. Without trunking, you only have the single port thing. It is quite easy to open the correct ports for SIP, some just have GUI with a SIP checkbox, IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

://en.wikipedia.org/wiki/Venice to describe the area where Jews http://en.wikipedia.org/wiki/Jews were compelled to live. On Tue, Feb 28, 2012 at 6:55 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, but

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Roger That! On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy

Re: [asterisk-users] [asterisk-dev] SIP, NAT, security concerns, oh my!

handy. That is my approach until IPV6 ever come out, or some other variant. Thanks, Steve Totaro Thanks, Steve Totaro I use a lot of Zentyal for OpenVPN plus networking fun.  I did hear from a snom engineer that they got the openvpn working with a limited functionality on the snom 300

[asterisk-users] **OT** Fwd: oFono 1.0 has been released

When oFono launched, I announced the project to other projects that it may compliment. oFono has hit the 1.0 Milestone and has some serious backing if you missed my post a year or so ago and never heard of it. Check it out... Thanks, Steve Totaro -- Forwarded message -- From

Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

Going with the flow on top posting. You just need an SBC http://www.voip-info.org/wiki/view/Session+Border+Controller setup correctly. Thanks, Steve T On Sun, Oct 2, 2011 at 3:14 AM, Sam Govind govoi...@gmail.com wrote: Hey, Why do you think using OpenSIPs is not going to work for you ? You

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

On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett adamli...@plexicomm.netwrote: 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 problems sending or receiving faxes?

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. If

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons

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

I gu On Thu, Aug 25, 2011 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Aug 25, 2011 at 12:39:14AM -0400, Steve Totaro wrote: I used the Asteirsk System() app to call lynx with a special URL. The URL contains all the authentication, recipient, and SMS body. Calling

Re: [asterisk-users] SIP client on a mobile?

On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls. This thread morphed from a nice home phone system into something completely different. Yup

Re: [asterisk-users] SIP client on a mobile?

On Thu, Aug 25, 2011 at 3:06 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls

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

So in other worlds you had nothing to contribute to this thread. On Thu, Aug 25, 2011 at 2:44 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: VoIP mostly aside, a couple more thoughts. I am not sure I understand your reasoning for DISA or how it is cheaper. The only

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

On Thu, Aug 25, 2011 at 4:36 AM, Skyler skchopper...@gmail.com wrote: Steve, On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote: ... For fax, I use Hylafax and for text, I use Kannel. These are WAY more powerful than Asterisk apps. With Kannel, I used the Bluetooth GSM modem

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

On Wed, Aug 24, 2011 at 2:42 AM, Per Jessen p...@computer.org wrote: Linuxguy123 wrote: My original post didn't mention it, but I would like my home system to be Asterisk based. Has anyone figured out how to minimize cell charges when on the road via making calls via the home phone

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

NBX100 by Polycom. True plug and play. Can be IP but uses MAC on the LAN except for the PBX. It may be discontinued but there is plenty on Ebay. Cheap, scales well, tons of options. On Wed, Aug 24, 2011 at 10:34 AM, C F shma...@gmail.com wrote: The 824 is NOT discontinued. On 8/23/11,

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