Re: [asterisk-users] Investigating international calls fraud
You don't mention if the phone is remote, or local. Although you do mention it had a default user/pass. If the UI of the phone was/is accessible from the I'net, the GUI does have the ability to place a call from it, that is one way the calls could have been placed. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven McCann Sent: Wednesday, January 28, 2015 4:03 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Investigating international calls fraud Hello, I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the phone company (CenturyLink in Texas). Some details: * PBX is located in Texas * Phone carrier is CenturyLink * FreePBX distro running asterisk 1.8.14 * source SIP extension is Mitel 5212, firmware 08.00.00.04, default admin password (argh!). Phone is used by many different people. More PBX setting details: * inbound SIP traffic is not allowed through the firewall * internal network is not accessed by many * FreePBX web interface Questions I have at this moment: 1) how were the calls placed? Was the Mitel SIP phone hacked somehow? Asterisk PBX? 2) how does this typically get sorted out with the phone company? they are charging $6.25 per minute for the Texas to Cambodia calls. The phone system owners are at fault, but how have these situations worked out in the past? I'll be tightening things up, but any feedback is appreciated. Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, September 18, 2014 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications Eric Wieling wrote: You don't mention your endpoint Both ends of the PRI. In house the end points are SIP phones, but calling from a sip phone (Polycom) to our remote office, there is no ringing. I'll be on site again this Saturday. I may end up putting the old 1.4x box back into place, I did get it working again. Doug A user on the PBXinaFlash forum has the exact same problem, albeit on a different card http://pbxinaflash.com/community/index.php?threads/no-inbound-dahdi-progress-ring-need-help-adding-playtones-ring-properly.15478/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling [ewiel...@nyigc.com] Sent: Thursday, March 13, 2014 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto, bottom posting is from the 90's. We've passed that era. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need a second opinion on a new phone systemdeployment
Another option instead of 2 servers dedicated as PRI gateways is to use AudioCodes Mediant 1000 or 2000 gateways. Either of them will also failover to a backup proxy if the primary proxy (server) is offline. Probably much cheaper than the kick ass box you plan to build + PRI card(s). I'm not affiliated either, but we do place them in our 911 call centers. They have analog gateways as well for FXO FXS devices. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, June 14, 2013 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need a second opinion on a new phone systemdeployment 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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Voip Trunking
E911 does not follow the standard SIP RFC. That would be a good reason that they couldn't/wouldn't do it. Now that I say that I should qualify it and say NG911 (or ESINet) does not follow SIP RFC http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your county is not using standard SIP for E911, it just wouldn't be considered NG911. From: Chris Nighswonger Sent: Fri 4/19/2013 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] E911 Voip Trunking During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8.13.1 -- how to limit voicemail emailswhen the caller hangs up before they leave a message?
Sounds like you need disconnect supervision enabled somewhere. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Piszcz Sent: Thursday, November 01, 2012 11:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk 1.8.13.1 -- how to limit voicemail emailswhen the caller hangs up before they leave a message? Hello, I use asterisk with an SPA3102 (latest F/W). I have my asterisk 1.8.13.1 voicemail.conf setup as follows: ; Limit the minimum message length to 3 seconds minsecs = 3 This works perfectly, however, when the caller hangs up before the beep (or during it?) then I get 1 minute and 22 seconds of (3-5 sec of dialtone, then saying to dial the operator)). How do I avoid getting this? If a message is not left, I do not wish to receive any e-mail/attachments like this, are there any workarounds? I assume this may be related to the SPA3102 but am curious to learn how others deal with this problem/if they have this issue. Name: Voicemail Message Number: 5 Mailbox: 1 Caller ID: S X Caller Name: S XXX Caller Number: X Duration: 1:22 Date: 20121101_1116 The voice mail: http://home.comcast.net/~jpiszcz/20121101/msg0004.WAV Justin. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN termination in Virtualized AsteriskEnvironment
Or Audiocodes, or MediaTrix, or … From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Thursday, May 31, 2012 3:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; a...@avhan.com Subject: Re: [asterisk-users] PSTN termination in Virtualized AsteriskEnvironment You need to look at Redfone fonebridges to achieve this. Please connect with me offline, we have it working in India in our CloudVoice Infraatructure. Mitul Limbani On May 31, 2012 12:40 PM, Amit Patkar | ATPL a...@avhan.com wrote: Hi Lot of users have deployed Asterisk in virtualized environment like VMWare, KVM, Hyper-V. Where as can we use Digium / Sangoma PRI cards in virtualized environment? If yes, then How? What kind of configuration is required? If not, then how is PSTN termination achieved in virtualized Asterisk deployment? Thanks Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
This thread may interest you. Add a SSD and RAM and you're good to go! http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200. 12460/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, May 10, 2012 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Link2VoIP going out of business! Now what?
Voip.MS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Royce Souther Sent: Monday, March 05, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Link2VoIP going out of business! Now what? Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A new hack?
Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Saturday, November 26, 2011 6:55 AM To: Asterisk Users Mailing List Discussion Subject: [asterisk-users] A new hack? Or just an old one that I've not noticed before... Seeing lines like this in the logs: [Nov 26 08:47:17] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=E2lb2p9BOJ [Nov 26 08:47:17] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=XMDRarBM2w [Nov 26 08:47:19] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=AaTE0L0oRj [Nov 26 08:47:21] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=igsN240Wr5 [Nov 26 08:47:23] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=E8Nkbs0Aye [Nov 26 08:47:25] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=LEvpc7tK6B [Nov 26 08:47:27] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=WrIoZ92YPz [Nov 26 08:47:29] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=kuGTjXr7Pd [Nov 26 08:47:31] NOTICE[789] chan_sip.c: Sending fake auth rejection for user VOIP sip:VOIP@85.25.145.176;tag=ygQBLSjH1m etc. The IP address is presumably the IP address of some compromised host (in Germany in this case, but I've noticed others around the globe so the software doing it would appear to be widespread) - it's not a host that should be connecting in. I supect that some SIP PBX somewhare is vulnerable to having an account called VOIP, so this remote attack is trying to compromise that account. At least it's only once every 2 seconds, so in that respect no worse than the multitude of pop/smtp/imap/ssh type attacks that hackers try... I've seen it on several servers now, always for account VOIP. I'm presuming the fake rejection is the side-effect of using alwaysauthreject in sip.conf. (if-so, then it's doing the right thing) But something to look out for just in-case.. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Integration with Android device
sip.conf, look at externalip, externalhost, and localip. From: Gopal krishnan Sent: Wed 8/24/2011 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
If this is what you need (fax/SIP/SIP Trunking/Vmail to email/Fax to email) and are willing to run on real hardware, or a virtual machine (not an embedded device), look in to PBX in a Flash along with IncrediblePBX/IncredibleFAX addon. This setup will do everything you want, and then some. It may take you a few weeks to setup and tweak to your liking, but once it's up and running, you won't look back. I cut my phone bill from $58/mnth for POTS with a few services to ~$4/mnth with everything under the sun. That savings has enabled me to buy SIP hardphones for around the house, and of course every laptop, netbook smartphone has a client installed. We both have DISA enabled and our access line programmed as a MY5 number, and we route all of our calls through the server at home. My cell bill went from ~$100/mnth to $48.50. Don't skimp on the hardware, use a pc (any old P4 with 512-1G of RAM will do), embedded devices just don't have the horsepower to make a featured Asterisk server shine. From: Linuxguy123 Sent: Wed 8/24/2011 10:49 AM To: skchopper...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system On Tue, 2011-08-23 at 21:36 -0700, Skyler wrote: Hi, On Tue, 2011-08-23 at 21:50 -0600, Linuxguy123 wrote: So you have asterisk loaded on a wireless router ? Linksys 54G by chance ? Yes, Asterisk at the moment. Cisco E3000. 54G is too small for asterisk, not enough flash/cpu. OK. I'm a 54G guy. I just bought a E4200 the other day for our media network. Which VOIP phones are you using ? Which ATA are you using ? I have Aastra 6731i, PAP2T, HT286 a Polycom and an Snom unit. Linphone, Bria, jitsi work as well for PC/Mac/iPhone. Any voip device/software would work. OK. The wife uses call-through on her Blackberry with MY10, she adds contacts with a pause after her voxnumber; like 1NPANXX,personsnumber so it dials in then dials out on the trunk. We have unlimited 60 countries so we can literally call anywhere, from anywhere and never have to think about it. Our plans have free local calling and 20 cents a minute for long distance. I spent $80 last week on long distance that would have been $12 on our home plan. To say nothing of all the other benefits. Took me 6 months here-and-there to get it this far. Well worth it though as we save about $180/month in cell phone bills now between us. Right. Are you using a POTS connection or SIP provider for your phone system ? How big is the system ? (number of lines, users, etc.) Just family and tinkering. I had load tested it with SIPp simulating 10 concurrent calls, sat at a steady 93% cpu. I'd say the E3000 would suffice for home use, 2-3 concurrent users. We stream off the NAS through it also and don't even notice during a call. Sounds perfect. How does a wireless router handle voicemail ? Ie no hard drive, so where does it store it ? NAS ? It records to memory (flash) and sends a wav to email. Fax works the same way. :drool: So you can receive faxes that arrive at home on the road then, as an email attachment, right ? Without having to find a fax machine while traveling and coordinating with the sender ? If we wanted faxes received on the fax machine, can asterisk recognize a fax tone and route the call to the fax machine ? Will the fax machine send via an analog connection to the asterisk system ? Or does it need its own line directly out ? What information resources did you use when setting up your system ? Thanks again for the replies. LG -- _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hide google voice number
Voip.ms actually offers more features. Depends on your needs. I use both as long distance carriers. My DID's are from Voip.ms. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.H. Jos Sent: Friday, July 29, 2011 4:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Thank you Terry, CallWithUs is what I am looking for, the most feature rich VoIP service!!! I hope it will not be difficult for me to have it working with Asterisk and OpenBTS (It's worth to see what OpenBTS is) On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell te...@brummell.net wrote: Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other VTSP's) do. From: A.H. Jos Sent: Thu 7/28/2011 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Do you mean that was possible to set the CID in the early days of GVoice? On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.net wrote: Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. From: A.H. Jos Sent: Thu 7/28/2011 9:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hide google voice number Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hide google voice number
Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. From: A.H. Jos Sent: Thu 7/28/2011 9:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hide google voice number Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hide google voice number
Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other VTSP's) do. From: A.H. Jos Sent: Thu 7/28/2011 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Do you mean that was possible to set the CID in the early days of GVoice? On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.net wrote: Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. From: A.H. Jos Sent: Thu 7/28/2011 9:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hide google voice number Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Asterisk Box was hacked
Really, since you sound like a novice in the Asterisk world, maybe rolling your own solution isn't a good idea. Why not use an all-in-one solution like PBX in a Flash? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito Sent: Thursday, July 21, 2011 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] My Asterisk Box was hacked Hi List, My asterisk box was hacked! Can anyone help on how do I secure my asterisk box, currently my box is installed with 2 NIC. 1st NIC is for LAN access and 2nd NIC has a public IP which is registered to our VoIP Provider. As I remember I already tried putting our Box on NAT but unfortunately due to some issue like call is dropped after 30 seconds and sometimes voice are not heard. Then we disable again the NAT. Your advise will be much appreciated. Thanks in advance. Regards, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributing the incoming calls and the huntgroup
FreeBPX calls them Ring Groups, you can look in to that. Or you could use a small ACD group. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Saturday, July 02, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Distributing the incoming calls and the huntgroup Hi All; To be able to distribute the incoming calls on a group of extensions, is there huntgroup in Asterisk? Or what I have to use? I need first call to be send for extension 500 and second call to be send for extension 501 and third call to be send for extension 502 and fourth call to be send again for extension 501 and so on .. I searched for huntgroup in Asterisk, but did not find any thing related to huntgroup in asterisk ! It look like there is not huntgroup in asterisk?! So how to distribute the calls? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Once you set it Advanced, you can't see how to do it, so I can't tell you. It's under one of the menu's, somewhere... From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
R7.2.07.02.00.04 And yes, that is likely the cause. From: vip killa Sent: Wed 6/22/2011 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. From: vip killa Sent: Wed 6/22/2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.net wrote: R7.2.07.02.00.04 And yes, that is likely the cause. From: vip killa Sent: Wed 6/22/2011 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Yes. From: vip killa Sent: Wed 6/22/2011 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Do you have BLF working on the Mitel? On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com wrote: Ahh then it makes sense, FreePBX checking to see if the line is in use, then sending busy signal instead of interrupting the call On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.net wrote: PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. From: vip killa Sent: Wed 6/22/2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.net wrote: R7.2.07.02.00.04 And yes, that is likely the cause. From: vip killa Sent: Wed 6/22/2011 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
Might take a bit, putting out some fires and beating back some aligators here at work, but I'll get it to you. From: vip killa Sent: Wed 6/22/2011 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Any chance you could send me (off list) you're example provisioning files (without the SIP credentials and IPs of course)? I can't find them anywhere online. On Wed, Jun 22, 2011 at 11:21 AM, Terry Brummell te...@brummell.net wrote: Yes. From: vip killa Sent: Wed 6/22/2011 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Do you have BLF working on the Mitel? On Wed, Jun 22, 2011 at 10:36 AM, vip killa vipki...@gmail.com wrote: Ahh then it makes sense, FreePBX checking to see if the line is in use, then sending busy signal instead of interrupting the call On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.net wrote: PIAF with * 1.8.3 My bootrom is 2.3.2.2 also. From: vip killa Sent: Wed 6/22/2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 i just upgraded to R8.0.08.00.00.04 but i can't find the bootrom upgrade so i'm still running 02.03.02.02 tested and call is still being interrupted when paging it... are you running straight asterisk or is something else handling the dialplan when you test? On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.net wrote: R7.2.07.02.00.04 And yes, that is likely the cause. From: vip killa Sent: Wed 6/22/2011 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Hmm, could be im on old firmware but i don't see SIP Enhanced Mode and i followed instructions in that PDF. would you be able to tell me what firmware you are running? On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote: http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf Page 32 From: vip killa Sent: Wed 6/22/2011 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 How do you set them to Advanced SIP mode? On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote: My Mitel sets are all in Advanced SIP mode (I think that's what the call it), have you done this? Once you change to Advanced SIP, you can't go back to basic SIP. From: vip killa Sent: Wed 6/22/2011 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call paging interrupts call when using Mitel 5224 Thanks, that must mean it's not asterisk but the AGI/AMI software we use along side it. On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
I have a 5224 and 5220's, I will try it tonight when I get home. From: vip killa Sent: Tue 6/21/2011 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call paging interrupts call when using Mitel 5224
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Tuesday, June 21, 2011 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call paging interrupts call when using Mitel 5224 Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. This does not happen to me, my call stays up. Caller with the page gets a busy signal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re : Re : Direct RTP with Asterisk
I didn't think it was possible if the endpoints, or Asterisk was behind a NAT. Someone please correct me if I am wrong. http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup From: Sagbo Romaric Sent: Sun 6/19/2011 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re : Re : Direct RTP with Asterisk I want to build an architecture with client behind NAT without VPN. With can reinvite, the RTP doesn't go directly, I have something like one way audio. Best, De : Roger Burton West ro...@firedrake.org À : asterisk-users@lists.digium.com Envoyé le : Dim 19 juin 2011, 15h 24min 07s Objet : Re: [asterisk-users] Re : Direct RTP with Asterisk On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote: No, I can't, because, it's a different NAT. I try to simulate P2P with asterisk. What you suggest to me ? I like VPN tunnels. They give you a flat network topology and decent security. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issue
I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you have a message. You press 1 to play and she just says First then gives you options to delete, save etc. The message is in the INBOX as msg0001.wav currently. From: Alec Davis Sent: Wed 6/15/2011 4:12 AM To: brya...@zktech.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Voicemail issue https://issues.asterisk.org/jira/browse/18998 may have the answer, particularly the patch bug18998-1.8.2.3.diff.txt Alec From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, 15 June 2011 12:11 p.m. To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail issue Ok here is a step by stop on how I can repeate the stuck voicemail box bug. Now how do I fix it? Again I am on version 1.8.2.3 build. Can some one with a newer build test and tell me if they get the same results? Example: All testing has been done with a single message in the inbox. User has a message in their inbox They call in they listen to the message. They press 9 to save the message. They select to save the message back to the 0 folder (inbox) The system changes the messages index from to index 0001 The user hangs up The system leaves the message as index 0001 The user calls in again and it says they have messages but because there is no index so they cant get at any messages in that folder. This explains over 50 instances where voicemails would get stuck in boxes with no indexed message. How do I fix this issue ASAP? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Bryant Zimmerman brya...@zktech.com Sent: Tuesday, June 14, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail issue Hey all I am having instances where voicemail boxes will have a 1 message and no 0 message this causes the user to be told that they have a message that they can't get at. If I renumber the messages manually to start with the 0 numbering then the user can get their messages. What could be causing this and how can I get it out of the system. Is there a patch I can apply to the system or is there a know fix for this issue. Right now I am stuck on this version because of some bugs in the current release that are show stoppers. I am on 1.8.2.3 build. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ground Start ATA / VOIP Gateway
From: John Novack Sent: Tue 6/14/2011 3:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway Robert Huddleston wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. I don't know of any ATA that will do GS An RJ-21 is the designation for a 66 block with 25 pair connector on the side GS is available with many channel banks though a T1 card and channel bank might be overkill for your application. Is this to go into a legacy switch? Most have line cards that can be easily switched to Loop Is this in the US, or ??? John Novack -- AudioCodes make Ground Start FXO's if that's what you are looking for. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Sent: Sunday, June 12, 2011 1:51 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] A question about Caller ID Hi, over anlog lines. Many thanks, Christian Bell 202 modulation between the first and second rings... Read the operation section from the following link. http://en.wikipedia.org/wiki/Caller_id -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from Mantis to JIRA
We use Jira at work. I hate it. Hope you have a better experience than I've had! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Bryant Sent: Wednesday, June 01, 2011 7:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Migration from Mantis to JIRA Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be: https://issues.asterisk.org/jira/ Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org. IMPORTANT NOTE: You will have to click the forgot my password link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm. For more information about how to use JIRA, see the JIRA user's guide: http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide If you run into any problems after the migration has taken place, please report them in the JIRA Help project. If you would rather report something via email, email espiceland at digium dot com and me. Thanks, [1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
You are doing a CNAM lookup on that 202 number. Change the URL to a number you know, and it will do a CNAM lookup on it. You can take your tinfoil hat off now. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Sunday, May 29, 2011 8:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Free CNAM BE WARY OF THIS ONE! If you click the link it comes up with a simple block Text Message US GOVERNMENT I doubt the US Government has any thing to do with it but... something is fishy here. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R. Wally Sent: Sunday, May 29, 2011 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Free CNAM FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 You can monitor the stats, including the current queue size, at freecnam.org API Results will continually improve as the database grows, so please be patient with limited results at this early stage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] standalone PRI-to-SIP converter
From: Patrick Lists Sent: Fri 5/27/2011 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] standalone PRI-to-SIP converter On 05/27/2011 05:10 PM, Michelle Dupuis wrote: I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Have a look at Patton or Audiocodes. Both solid solutions although Audiocodes has a bit of a reputation when it comes to configuring it. Regards, Patrick -- I have a 2nd vote for AudoCodes as well. We use Mediant 1000's (combo FXS, FXO, PRI, T1/E1) and Mediant 2000's (digital links only) boxes here. They can be a bear to set up as they try to please everyone for everything in various setups. They have a ton of config options, but once they are up and running, they are rock solid. I mean ROCK solid. The price point may drive you away though. Case in point, we needed to do verification on a fully equipped M1K (Mediant 1000) to certify it to work with our product. The boxes we ordered have a PRI/T1 module, FXS, FXO Loop Start, FXO Ground Start, and dual power supplies. As ordered the boxes were somewhere in the neighbourhood of $7000 each. We use AudioCodes exclusively with our product, MP-11x's, M2K's M1K's. So I'm quite used to them, but for a newbie they are VERY scary! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping incompatible voice frame
For 2 different hosts. SIP/voxbone.com and SIP/4420 From: RSCL Mumbai Sent: Thu 5/19/2011 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial from voicemail
From: Kelly Opal Sent: Tue 5/3/2011 1:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dial from voicemail Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail or to just leave a message at the tone. Thanks Kelly -- _ Any FreePBX based distro does this, so it is possible. I have no input on how to program it on vanilla * though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No messing around with switches and cables an crap. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Saturday, April 30, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HA Asterisk Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T auto answer?
No. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, April 25, 2011 6:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PAP2T auto answer? Hi all, Is it possible to send a SIP header to a PAP2T or SPA and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unresponsive
http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory From: Jonas Kellens Sent: Mon 4/18/2011 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk unresponsive Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples Asterisk freezed and only a reboot of the whole server fixed this. Any command on the Asterisk CLI was not executed because Asterisk was too busy processing all of these messages that you see in the debug log. What is the origin of these messages ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration from '000000 x 1000
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Saturday, April 02, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Registration from '00 x 1000 On 04/02/2011 02:08 PM, Steve Davies wrote: On 2 April 2011 09:46, Jonas Kellensjonas.kell...@telenet.be wrote: Hello list, I often see the following in my message log : [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found And there are hundreds of them... Is there a setting so I can make Asterisk not respond to SIP PEER registrations which are not in my sip.conf or my realtime MySQL DB ?? Yes, you add a rule to your firewall! Even better, get it filtered further out so that it does not waste your inbound Internet bandwidth, because in my experience, once those SIP spammers start, they continue for weeks at the very least. IIRC, the way SIP registrations works basically requires than an failed/un-authorised attempt is responded to, so that the other party knows to authenticate. If you stop sending that response, no-one can authenticate. Hope that helps. Steve So in short, there is no way of throwing away registrations that are not in sip.conf. The only thing I can do is check the messages file now and then to see if there were bad registrations, and then blacklist them. Kind regards, Jonas. Search the archive for Fail2Ban, it is what you are looking for. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Your delay is due to the amount of time the F2B script takes to read the log file, and due to how often it is called. I do not believe it is a realtime event. Say, every minute it's called to read the log and act. I'm not sure of the exact numbers, but you get the idea From: vip killa Sent: Thu 3/31/2011 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban Back to the original question, for those of you using Fail2Ban, Does it take an unusually high amount of break-in attempts before attackers are banned? I have it set to 5 attempts in fail2ban but usually, the attacker is able to make over 100 attempts before fail2ban bans them. I've tried this using asterisk's /var/log/asterisk/messages and /var/log/messages with same results. Perhaps someone else is experiencing this or has resolved it, thank you. On Thu, Mar 31, 2011 at 4:05 AM, Gordon Henderson mailto:gordon%2baster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: Yah, sounds simple, how do you set it up to do this? Fail2Ban was pretty easy, if it's that easy, why was F2B even created? It's easy for me because I read an undestand how things work, and deal with Linux firewalling in a daily basis. Fail2ban is an (almost) drop-in solution which requires minimal thinking - just a few lines in a config file to edit. (and python which I don't have installed on my systems) Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
I think you will find Fail2Ban the defacto standard. From: vip killa Sent: Wed 3/30/2011 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban so does anyone use fail2ban w/ asterisk or most people use sshguard? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, March 30, 2011 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban could you please elaborate on how you have iptables setup to work that way? On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson gordon+aster...@drogon.net mailto:gordon%2baster...@drogon.net wrote: On Wed, 30 Mar 2011, Terry Brummell wrote: I think you will find Fail2Ban the defacto standard. I don't use fai2ban. Never have, never will because I simply don't need it. Standard iptables are good enough if you can be bothered to use them to their full abilities. No need for anything else as iptables can do connection tracking and blocking against time - just like fail2ban does. More than X connections a second/minute/hour from a given IP address? Yes, iptables can detect and block that. Works for all protocolls too - SIP, IAX, POP, SSH, etc. Gordon -- Yah, sounds simple, how do you set it up to do this? Fail2Ban was pretty easy, if it's that easy, why was F2B even created? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Provider Recommendation in US
VOIPo, CallCentric, F9, Voip.ms, CallWithUs to name a few (no particular order, just what popped in my head) From: asterisk-users-boun...@lists.digium.com on behalf of Brent A. Torrenga Sent: Thu 3/3/2011 11:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Provider Recommendation in US I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [zapata.conf] What is wink?
http://www.carrieraccessbilling.com/telecommunications-glossary-w.asp From: asterisk-users-boun...@lists.digium.com on behalf of Gilles Sent: Tue 3/1/2011 7:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] [zapata.conf] What is wink? Hello I couldn't find information about what wink is in zapata.conf: www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters Does someone know what it is, and how it differs from flash? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk security....again
When he says customers I am assuming he means remote customers. It sounds like he is a reseller of telecom facilities to me. Which means his customers most likely have ATA's with port 5060 forwarded to the ATA, or they are direct on the I'net. He has already set the ATA to only allow calls from the proxy server, so sounds like he has plugged the hole. They are not 'sniffing' your traffic, they are guessing/scanning. That's it, that's all, no great conspiracy going on. They look for open 5060, then send SIP requests to it hopefully finding a badly implemented SIP solution to which they can dial through. Once they determine they cannot get through, the script will move on to the next sucker. You have a couple of options, which you could implement at *each* of your customers if you wanted. Set up a VPN, tunnel the SIP/RTP traffic through it. Set up IPTables at the customer to only allow SIP from your IP. Or, do what you have already done and forget about these idiots doing the scan, they are harmless at this point. Vlans and DMZ for the server do no good as the attacks are being directed at the remote client side, not the server. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Monday, February 28, 2011 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk securityagain Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. The IP address of you server can be revealed in some unincrypted SIP signaling of some call through the Internet to/from your server's client, or simply by your client SRV record in the DNS, if you added it to his DNS. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. Regards, Ricardo Carvalho. On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com http://www.axvoice.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
If you compare a working config with a non-working you will see something with the answer type. I had that issue until I down rev'd. Look for something like Ring Answer, I forget the exact details now. From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 1:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Paging with Polycom 3.3.x Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer correctly. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike Looking for it now -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Thursday, February 17, 2011 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier Cintrón Olguín Sent: Thursday, February 17, 2011 7:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Yes, I use Elastix myself too. Funny that I didn't mention that one! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Friday, February 18, 2011 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Thursday, February 17, 2011 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier Cintrón Olguín Sent: Thursday, February 17, 2011 7:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Yes, use a FXO device, like the AudioCodes MP-114. It is an external gateway that will allow you to interface your PSTN lines to Asterisk via IP. There are other brands out there but in my line of business we only use AudioCodes. From: asterisk-users-boun...@lists.digium.com on behalf of Francisco Javier Cintrón Olguín Sent: Fri 2/18/2011 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... Is there another way to interface to 3 external and 6 internal lines?? Thank you for your kind help winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
Aastra Polycom because they can be configured using a TFTP server. Great for large installations with centralized management. Mitel 5215/5224 because they are dead simple to configure (via web gui) and just plain work with no screwing around. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ast guy Sent: Saturday, February 12, 2011 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Hardphone that works well with asterisk Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know, the Mitel's do not support tftp/http provisioning. I did just upgrade my 5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't know what the phone is trying to do in that folder. Anyway, that's taking this off topic of the OP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Saturday, February 12, 2011 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Hardphone that works well with asterisk Terry Phone Provisioning is a part of Asterisk. It works over HTTP now and with an FTP or TFTP proxy can work over multiple protocols at once. Read More: https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk I added example snom support and will have to start a review board for adding Cisco, Aastra and others. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
