Re: [asterisk-users] doorphone?
On 03/09/2011 02:57 AM, Dan Journo wrote: could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I would recommend using a normal doorphone, and connecting it to a SIP gateway like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo That's not always true. Some door phones have a remote unit that connects to the network and a local device at the door, giving some better security. I've used the Valcom VIP-172 phones. They are simple and work well. Very good support if you need to call them. http://www.valcom.com/Home_links/sipdoorintercom.htm Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[asterisk-users] AstLinux 0.7.5 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.5 with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More information about the release is available on our website: http://www.astlinux.org/content/astlinux-075-release Direct links to the installation files are available here: http://www.astlinux.org/release/075-asterisk-1811 http://www.astlinux.org/release/075-asterisk-1436 All current users are encouraged to upgrade to one of those releases. A firmware upgrade can be performed from the web interface or from the command line. Command line upgrade: (for Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware (should report astlinux-0.7.5) then upgrade-run-image upgrade http://mirror.astlinux.org/firmware or (for Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware (should report astlinux-0.7.5) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware If you are upgrading from an Asterisk 1.4 base to Asterisk 1.8, you will need to manually update any Asterisk related configuration files. Please ask any questions about this release on the AstLinux-user's mailing list. -- The AstLinux Team -- _ -- 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] Issue with MOH - Asterisk 1.4.17
On 12/03/2010 03:30 PM, Doug Lytle wrote: Napoleón Ernesto López Espinoza wrote: We're sorry, your call did not go through. Any clues about this issue? How about some output from your console when it fails? It's would also be advised to use a much more recent version. Asterisk 1.4.17 has many bugs and security issues that have been addressed in newer versions. 1.4.37 is the latest version from the 1.4 branch. It's quite possible that whatever you're trying to fix is already fixed in that newer release. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Asterisk choking on voice messages announcements
Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Using the PBX Directory from a Blackberry
On 07/02/2009 10:14 AM, Tzafrir Cohen wrote: On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure is there is actually tones generated when the alpha characters are pressed, it just doesn't work). Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Is it a tone? Or the letter itself in SIP / RTP signalling? This is a 'bug' or 'feature' of blackberry phones. The phones switch the keypad to numeric when in a phone call. You need to memorize abc=2, def=3... Sure would be nice if there was an option to send the DTMF for 5 when pressing the alpha key j k or l, but I don't believe this is possible. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If not, you should. On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice disappears after 1, 2 seconds. I'm trying to understand if it is a bandwidth problem. At the moment I have my phone connected to the openvpn client (which is its gateway) but I have to use the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip (192.168.1.12) is not working. I suppose it is a sip protocol problem: I had to change the sip.conf setting nat=yes to make the phone dial and domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). I keep on working on the vpn since it seems so little is missing to have a clear conversation. Let me know if your tests are successfull. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
Giorgio, tcpdump and wireshark are your friends. Instead of guessing, capture a call with tcpdump then look at it with wireshark. Darrick On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote: Hi Darrick, I always set canreinvite=no 'cause it gives a lot of problems if set to yes (and the default is). I made a call with rtp debug on and I noticed that normally, on the asterisk CLI, I see one packet sent corresponding to one packet got (made a test with a local call on our production server). On the other server with the vpn, I get a bunch of sent followed by a group of got...there is something in the way the RTP packets are sent/received by Asterisk and maybe it can be correlated to the missing audio. Giorgio Darrick Hartman (lists) wrote: Do you have 'canreinvite=no' in your sip.conf entry for this phone? If not, you should. On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice disappears after 1, 2 seconds. I'm trying to understand if it is a bandwidth problem. At the moment I have my phone connected to the openvpn client (which is its gateway) but I have to use the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip (192.168.1.12) is not working. I suppose it is a sip protocol problem: I had to change the sip.conf setting nat=yes to make the phone dial and domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). I keep on working on the vpn since it seems so little is missing to have a clear conversation. Let me know if your tests are successfull. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who has the clever Polycom upgrade system?
Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way to it. Can anyone point me in the right direction? That would be Karl Fife, of the famous Karl Fife experience. http://kfife.com/voip/ Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
Dean, I'm using Zabbix to monitor network interfaces, storage, cpu load and a few other things on several asterisk boxes. I'm just looking at adding Asterisk specific monitoring. Simple things like sip registration is pretty easy. Getting the actual status of zap-daddy hardware might be a little trickier. When I get something together I can pass it along. Darrick Dean Collins wrote: Has anyone ever 'released' an Asterisk module that is easily shared/downloadable? Or doesn't the nagios open source code work like that? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, 9 September 2008 9:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and Network Monitoring On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote: Dear Asterisk Users I'm looking for a solution that can be used to monitor Asterisk and the Telco lines aswell as the network (Servers, WAN LAN links, Router Switches) We use nagios for that. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interoffice phone setup
Michael Graves wrote: On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote: On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: What type of PBX hardware do you have on-site? Also what make/models of phones? Michael/Darryl, I do have a local asterisk box, which is why I am baffled. I am new to Asterisk and there is lots to learn, but my config is pretty basic, my sip.conf simply has the phones and single sip provider context in it. It doesn't make sense that the voip provider going offline takes the whole setup out with it. I am suspecting something else went south at the same time. I have snom m3's and one Astra 480i. Thanks! jlc I've seen this behaviour from Asterisk as well... while I can't say I have tracked it down and verified this... I've seen other talks about how Asterisk gets rather unhappy when it can't preform DNS queries. I suspect that may be your problem. Might want to check the archives for other issues that people have talked about DNS as a possible cause and see if there are any similarities. Yes, this is very true. Asterisk gets backed up trying to deal with lack of DNS. I'd diable the SIP trunk then restart. Perhaps this would permit internal calls to resume, as long as there are no attempts to dial external numbers. dnsmasq is just the creature for the job. Very flexible and easy to configure. http://www.thekelleys.org.uk/dnsmasq/doc.html -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell 1950
Eve-Ellen Cole wrote: I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students. Each time a voice mail is left, an email notification will be sent to the student. The email notification will provide a web link to direct the student to the voice mail itself. Anything I need to consider changing? I’m interested in any feedback you are willing to provide. Many thanks! If you're not restricted to 1U, I would strongly suggest getting more drives and doing a raid 10 array. Even if you are restricted to 1U, you can do a raid 10 array in the 1950, but you'll have to pay for the hyper-expensive 2.5 drives. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
Jonathan C. Bailey wrote: We've been using D-Link DES-3028P and DES-3052P switches. They can supply full power to EACH port unlike the Linksys switches we've tried. They're also rock solid from our experience. I echo that recommendation. The Linksys switches are probably the loudest that I've used. The D-Link's have been very reliable. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Alan Lord wrote: Lenz wrote: Hello list, after spending the best part of an afternoon trying to build Asterisk on an old EPIA VIA C3, I thought that writing a tutorial would make life easier for future compilers: http://astrecipes.net/index.php?n=356 I had never compiled Asterisk for a different architecture, and I'm pretty disappointed at how complex it is - building Zaptel, Libpri and Asterisk requires discovering three different procedures, and even passing the required architecture to the autoconfig module was not enough for a clean build - libpthread and libresolv would not link, so you have to add them manually. Aybody got an idea of who should be notified of this immediate problem, apart for the time-wasteful general compilation procedure? Thanks l. Hi there, I didn't find it too much trouble in a Via C700N system. But I wouldn't use one of the mainstream distros for the OS. They chew up system resources just trying to accommodate any hardware. The solution is to roll-your-own. See this series of articles on my blog... http://www.theopensourcerer.com/tag/asterisk/ The C7 supports full i686 features. The C3 is an older chip that is fully i586 and partially i686 compatible. If you have a distribution that is compiled with i586 optimizations, you won't have problems. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
David Nedved wrote: --- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still experiencing the same problem (not recognizing DTMF on SIP inbound calls) as well as new problems. The new problems are much more severe than the previous problems so I'm starting a new thread with a more descriptive subject. I've changed sip.conf to eliminate warnings for new syntax: insecure=port,invite dtmfmode=rfc2833; Choices are inband, rfc2833, or info Everything else is as-was in sip.conf, extensions.conf, iax.conf, rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked through the new samples and didn't see anything glaring I needed to change). For the config files I had not changed I took the new sample files. There were several things that changed... Now in addition to not recognizing DTMF on SIP still, asterisk is now frequently dropping calls when I start to enter DTMF. On console I get lines such as: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/x-081ea720, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(SIP/x-081ea720, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/x-081ea720, /home/dnedved/hello) in new stack -- SIP/x-081ea720 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'SIP/x-081ea720' status is 'UNKNOWN' Try adding this line in the general section of extensions.conf autofallthrough=no The default behavior in 1.2 was no. In 1.4 it changed to yes. That will be your simplest fix (without seeing your dialplan). Asterisk is moving on to the next step in the dialplan before you enter your digits. You need to have it wait for the digits to be entered. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF suddenly stopped working on SIP channel
David Nedved wrote: Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them what they changed (fat load of luck getting that question answered anyway). Everything was working fine with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6 valid combos of those two settings with no change. This is on asterisk 1.2.27 that's been working fine in production for about 3 months now. Here's the section from sip.conf (the way it had been working all along): [viatalk] type=peer secret=(yep it's right) username=(yep it's right) host=newyork-1.vtnoc.net canreinvite=no insecure=very qualify=yes context=incoming-viatalk dtmfmode=inband ; Choices are inband, rfc2833, or info ;relaxdtmf=yes ; Relax dtmf handling Thanks in advance for any help. I've got all incoming calls on Viatalk shunted to an extension in the meantime, not an elegant solution. Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardphone SIP phone costs
John Faubion wrote: are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and Out of curiosity, how do these phones handle the transition from Wi-Fi to GSM? Is it seamless? Can the transition occur when on a call? Not seamless unless the cell phone provider offers such a service. You won't find that available in the US. So even though it's one phone, you'd have 2 numbers. Cell phone providers have no incentive to offer such a hand-off because they wouldn't make any money on the calls after they are handed over to the voip system. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium registration utility version 3.0.3 released
Terry Wilson wrote: Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. Great! What will it take to get the g729 codec module compiled against uClibc? Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Declined: VoIP Users Conference
Doug Lytle wrote: Evan Ruff wrote: Since when is the users list a transport for calendar scheduling? Since when are humans infallible? Randy made a mistake. He apologized for it. Let's move on... -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
Tilghman Lesher wrote: On Friday 15 February 2008 23:53:19 [EMAIL PROTECTED] wrote: Practically any manufacturer gives similar support including ssh'ing in the users box. Really? Which manufacturers, specifically, will allow you to call up, get remote assistance, and help you get the card working like this? I'm not the person who wrote the original message, but I do know that Rhino will also ssh into a box if there is a problem. I started using Rhino cards because, at the time, Digium did not offer an analog card with built in echo cancellation. Most of my systems are single PCI setups (so even a separate echo cancellation card that requires a 2nd PCI slot does me no good). Apart from the occasional echo problems (which can be minimized with tuning), the TDM400p was a decent card. I'd buy Digium cards again but as a small business owner, it's much easier to stock replacement parts for one vendor (if someone's phone system goes down, they can't wait a day for replacement parts). I'd also like to know why you're posting anonymously, instead of standing behind your words. Yeah, that annoys the heck out of me too! If you're going to criticize or recommend something post with your real identity. That way people can make a reasonable conclusion on your advice. Who knows...perhaps the Shadow knows [1]. [1]: http://www.mysterynet.com/shadow/ -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
I told myself that I was going to stay out of this one, but since you find this important enough to reply twice to the mailing list with the same content, it must be worth my time to reply. If you carefully read the thread, the person who replied from Rhino went out of his way to NOT try to sell his hardware, until someone claimed that Sangoma is the best. I do not have first hand experience with Sangoma hardware. I am however one of the Astlinux developers. In that capacity, I can easily say that compiling the necessary modules for the Rhino cards is much easier than what is required to get the Sangoma stuff working. I have no doubt that in both cases the hardware is good. (I personally have not experienced significant trouble with Digium analog cards either, but they do take more time to adjust properly). Both hardware companies did work with us to ensure their cards will work properly with Astlinux. With the Rhino hardware, there is no need to compile extra utilities, only a zaptel module. The Sangoma setup is more complex. One plus in my mind is the Rhino card is made in the USA. I've also found the Rhino tech support to be excellent. Bottom line, use what works for you. I've used several Digium TDM400P cards and several Rhino analog cards. Both work well, but like I said earlier, the Rhino cards (with built in echo cancellation) were much easier to configure and get working out of the box. Darrick shadowym wrote: How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sando ma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNMP monitoring
Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.17 and Teliax DTMF
Carlos Chavez wrote: I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended for DTMF and Teliax? I've had no issues with our Teliax accounts since switching to 1.4.x. I stayed back on 1.4.16.2 so far because of one issue with parking calls. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
Tzafrir Cohen wrote: On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote: PC's age and when they age, things tend to go wrong, particularly when you upgrade software. Unusual crashes are usually the first sign that something is going wrong. And suddenly the same PC has unaged when reverting to 1.2? Again, you don't have enough data to be conclusive on that. So I humbly suggest that you won't be. It's more likely that there is something in the configs or dialplan that works fine in 1.2 but does not work well in 1.4. I have several machines that I migrated to Asterisk 1.4 that all are behaving just fine. My major motivation for moving to 1.4 was DTMF. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
bilal ghayyad wrote: Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? SSH uses tcp. Openvpn, by default uses udp. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Jared Smith wrote: On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote: Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? It's my understanding that SSH tunneling will only work with TCP traffic. IAX2 uses UDP packets, so I don't think that'll work. You might try setting up a VPN or something along those lines. (Also, IAX2 defaults to port 4569, not port 5060.) OpenVPN works great for this. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Checking that TDM card works?
Vincent wrote: On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote: wcfxo is not needed. Basically all you need is: modprobe your_card_s_driver This also pulls all of its dependencies (e.g: zaptel) modprobe wctdm Thanks, but on AstLinux, the modules are not unloaded: === pbx admin # /etc/init.d/zaptel stop pbx admin # lsmod Module Size Used by wctdm 31552 1 wcfxo 11424 0 zaptel188604 6 wctdm,wcfxo hdlc 22528 1 zaptel syncppp15300 1 hdlc ppp_generic28692 1 zaptel === Why would an init script not remove modules? Vincent, Come on over to the astlinux mailing list (on our sourceforge page). It will be easier to handle any Astlinux specific questions over there. But look in your /etc/rc.conf file for the ZAPMODS variable. You should have that variable set to: ZAPMODS=wctdm Beyond that, as long as Asterisk is not running, issuing service zaptel stop should remove all zaptel related modules. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Checking that TDM card works?
Vincent wrote: On Wed, 09 Jan 2008 06:01:32 -0600, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: But look in your /etc/rc.conf file for the ZAPMODS variable. You should have that variable set to: ZAPMODS=wctdm Yes indeed: #ZAPMODS=wctdm Should I add this module here, or in rc.modules? Uncomment that if you expect it to work. The module should be listed in ZAPMODS not in rc.modules. Are we positive that wcfxo is not needed in addition to wctdm and zaptel? Yes we're sure. Beyond that, as long as Asterisk is not running, issuing service zaptel stop should remove all zaptel related modules. Thanks, but it doesn't seem to unload the modules: == # /etc/init.d/zaptel stop # lsmod Module Size Used by wctdm 31552 1 wcfxo 11424 0 binfmt_misc11784 1 zaptel188604 6 wctdm,wcfxo hdlc 22528 1 zaptel syncppp15300 1 hdlc ppp_generic28692 1 zaptel slhc6784 1 ppp_generic # asterisk -r pbx*CLI stop now Disconnected from Asterisk server # ps PID Uid VmSize Stat Command (snip : no trace of Asterisk) 1327 root368 R ps # /etc/init.d/zaptel stop # lsmod Module Size Used by wctdm 31552 0 wcfxo 11424 0 binfmt_misc11784 1 zaptel188604 2 wctdm,wcfxo hdlc 22528 1 zaptel syncppp15300 1 hdlc ppp_generic28692 1 zaptel slhc6784 1 ppp_generic == I guess the zaptel script doesn't remove them, and I need to use rmmod manually? Since you have ZAPMODS commented out, the zaptel init script doesn't know which modules it should be using. I can assure you that this script does work properly if you have the configuration set correctly. Vincent, all of this is really Astlinux specific and would be better handled on that list instead. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone
Glenn Gillen wrote: Unfortunately there is only one port, clearly labelled handset On 31/12/2007, at 11:34 PM, dave cantera wrote: glenn, check your handset cord... it might be plugged into the wrong port in the back of the phone. perhaps the headset jack... daveC Push the cord all the way into the handset. I've seen some Polycom handsets that look like they are plugged in, but in reality, the end of the cord that plugs into the handset needs to go in farther. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
randulo wrote: Well, $9 would pay for up to 500 answers. I also found a free one I'm looking at now, but you never get anything really good free :) If $9 can put that survey together in a comprehensible set of questions and results, I will pay the $9. Let me see if I can put what you ask for together on the free one and post here in a bit. r Can you just install limesurvey on a server some place? It would allow you to do however many future surveys you want to do. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best cheap card to use for home Asterisk system???
Tim Reimers wrote: I have a single phone line (happens to be Charter Communications VOIP, but I have their ATA and they’ve connected to red/green pair in the house wiring) Ok. so they've installed an ATA which connects your analog phones to their VoIP (perhaps SIP) service. What I’d like to do is this: Get some low-end but reliable card/external adapter which would connect to their ATA and tie into Asterisk to take calls and faxes OK. Since we've established above that Charter's service is VoIP converted to analog, AND since Asterisk isn't really designed to work with fax over IP it is safe to say that it's not worth the effort to attempt to get this to work. I have relatives who have Time Warner's offering and even a stand alone fax machine will not work reliably over their internet phone service. Hell the audio quality is crap most of the time. I’m assuming this should be something with one FXO and one FXS port to connect the incoming line to and to connect the red/green wiring in the house to. I'm not sure if you're familiar with the Canadian television show that is popular on PBS in the US, but this sounds alot like the guy on the Red Green Show using duct tape to fix things. If you really want to use Asterisk, you'd be better off getting an account with a SIP provider and using an FXS adapter to feed line 2 on your phones similar to what Charter is doing with line 1. Linksys makes a decent adapter which would suit this purpose. Good luck! Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail playback on iPhone
Jason Lixfeld wrote: I guess what I'm asking is if there is a recipe anyone has used to allow a voicemail message (once recorded by asterisk) to playback on iPhone when said recorded voicemail is received as an email attachment. I understand you can convert using sox, so I guess that's the ingredient and some sort of * configs would be the glue - I suppose it's the glue I can't figure out. I'm not trying to figure out how to get voicemails to show up in iPhone VVM or anything like that. The iPhone can't play back wav or wav49 files? Check your voicemail.conf file. What format are you currently using? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
Kevin Smith wrote: Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. The major issue with this is most pc's are now coming with gigabit ethernet connections. Going to gigabit speeds is such a huge improvement it's often worth the extra expense to add a second drop to each location. Profiles will load faster, Outlook-exchange interactions work much cleaner. When gigabit capable phones are more prevalent, this becomes a non-issue. Right now, there are very few gigabit phones and none that are affordable. Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] net5501 + TDM400P?
Brandon Black wrote: Hi, I'm relatively new to Asterisk, and I'm looking to build a tiny system for home use. Welcome. What I'd like to do is set up an Asterisk box with 1x FXS (to the cordless phone base station) and 1x FXO (to the Vonage pseudo-PSTN-line), and also have it act as a SIP client to the office Asterisk system, and let the Grandstream be a client to my Asterisk, and configure it up so that both the Grandstream and the cordless system can be used to answer and dial from both my office and home lines. I'd drop Vonage, port your number to someone like Teliax and eliminate the voip to FXS to FXO conversion. What I'm considering hardware-wise is a Soekris net5501 paired with a Digium TDM400P equipped with 1xFXO and 1xFXS (might add a second FXS later down the road). I'm a pretty well-versed linux hacker, so I'm not too worried about being able to get the software side up and running. What I'm really looking for is advice and info from people that have tried this (or similar) hardware setups on the following: The net5501 would sufficiently support that load and then some. 1) Does the Soekris case fit a TDM400P card inside of it? I don't mind if I have to get out the dremel to get the ports exposed, but does it fit inside at all? Yes it will work fine. The case that comes with the net5501 will fit a half-length pci card. This is an improvement over the net4801 which would only support half-height/half-length cards. 2) With a net5501+TDM400P+1xFXS, if I use the Soekris 2.5A power supply, will I need the extra external 12V supply (the digium one) to supply power for the FXS module? Some things I read seem to indicate that you always need it, some don't...? Does anything change on this if/when I put in a second FXS port? With one fxs, you might be able to get away with it. You might look at the Soekris mailing list archives to see where you can pull some additional power if you need it. 3) How much storage space do I really need to get one of the common embedded Asterisk distros on there, and also leave room for a good amount of voicemail, and room to hack with cool things in the dialplan, etc? Would you recommend a 2.5-in HDD, or can we fit this all in compactflash? I don't really have any ideas on estimating voicemail space consumption, or commonly fun things people do with modules and dialplans that might suck extra space down the road. Have a look at Astlinux. http://www.astlinux.org The standard install is less than 64MB so you could easily get by with a 256-512MB compact flash card. Works well. We'd be happy to answer questions specific to Astlinux on our mailing list (hosted off the sourceforge project page). Note that the net5501 platform was added recently in trunk. None of the images have been released yet, but several of us have been running releases from SVN for several months. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best config for 12 FXO system?
Thomas Kenyon wrote: Tony Mountifield wrote: I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? No idea, but if you look further afield, if you buy a Sangoma A200 or an A400 you can have all 12 on one PCI (or PCI Express) slot (the former taking up 3 Spaces on your PCs backplane and the latter taking up only 1). If expandability is a concern, an A400 can support up to 48 FXO ports on one PCI (or PCI-Express) Slot (4 spaces) or an A200 can support up to 24 FXO ports. (6 spaces) I can't comment on how good they are, I've only got TDM400Ps myself. Rhino also makes some very nice cards and have a good support staff. The Rhino cards are also made in the US. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Doug wrote: http://www.atacomm.com/ ATACOMM Dear Atacomm Customers, We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm and its parent company Ataractic Corporation has ceased operations. We appreciate the 7 years of loyalty and support from our customers. We sincerely regret any adverse effects this may have caused. I'd say that's pretty self-explanatory. My credit card company is trying to recover about $800 in fraudulent charges for duplicate transactions and failing to send the merchandise for a transaction that dates back to late August. Normally I'd say this sort of thing belongs only on the biz list, but this sort of issue may affect so many people it's worth noting here (but not dragging out with hundreds of me toos). -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
Matt Watson wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. Matt, Look at his path. He's going from a PSTN phone to a g729 gateway. As long as the gateway is there, Asterisk doesn't really know about the PSTN phone. Therefore, yes, this should equate to pass through. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH Files Volume
Peder @ NetworkOblivion wrote: Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. run the files through sox -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Polycom 300/500/600
BJ Weschke wrote: On 8/31/07, Joe Acquisto [EMAIL PROTECTED] wrote: Any great disadvantage to using polycom 300/500/600 vs the 301/501/601? I recall reading in the release notes of the latest release of the firmware (2.2+) that I believe they've finally stopped supporting the earlier models so it looks like you are reaching or have reached an EOL period on firmware with those models. Aside from that, if you're happy with current functionality of those phones as they stand now, they'll probably be fine for quite some time to come. I believe they only dropped the 300 and possibly the 500 with the latest firmware (per the release notes for 2.2.0). The 600 already had the increased memory on the phone (which was the major difference from the 300/301 and 500/501. That being said, the 300, 500 and 600 are discontinued. You might be able to get them cheap, but I wouldn't use them in a new install. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Pico-ITX
Dermot Bradley wrote: Darrick Hartman wrote: Just because someone is using an old kernel or doesn't know what they are doing doesn't mean the hardware is bad. I've had very good success with dozens of different VIA boards (from the original mini-itx board up to current C7 models, the Jetway boards included). Not trying to start a flame war but I've had problems with kernels ranging from 2.6.16.19 through to 2.6.22 and I do generally think I know what I'm doing (long time Linux user). I've spent considerable time in the past 2-3 months trying to nail down the source of these lockups and have had no success. I haven't been able to get the boards to lock up. Since I don't think this is Asterisk related at all, let's take this off list. Email me your config. Also what distro have you been trying to use? Perhaps you've been beating down the wrong door looking for the problem. As I mentioned in a reply to Gordon last posting I have not tried C3 based systems so I can't comment on them. I'd be glad to exchange kernel config files with you for Jetway C7 systems as I really would like to the bottom of this - I have 3 J7F2 motherboards here that are useless to me unless I can find a solution. Would you be willing to email me the current kernel .config file you use for C7 boards? Go ahead and grab the latest image of Astlinux. There's a bootable iso image available so it should minimize the work you'd need to do to test this. In the past I have exchanged kernel configs with various people on the VIA Arena forums in an attempt to find the root cause but didn't see any obvious problem. Thanks in advance. Not a problem. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
Zeeshan Zakaria wrote: Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. I have a VIA C3 (PD1) system that I've been using for a few years. The C3 is a processor that started with Cyrix. VIA bought them out. The C7 processor is the successor to the C3. The C3 was discontinued because Intel would not renew a license agreement with VIA. The C7's have pretty much full i686 compatibility while the C3 is missing a few of the optimizations (it's fully compatible with i586). Astlinux has an image built specifically for the VIA boards even with some support for the Padlock feature (hardware crypto engine). These systems have performed quite well. They are low power and compact. Not sure what else you need to know. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? The fanless mini-itx boards should be just fine. There are too many factors to give you a definite answer, but I currently use one with a TDM400 card. A majority of the calls on the board are sip with no transcoding so there is a very small load on the system (hardly noticeable). If you are doing a ton of transcoding or recording calls, your results may be different. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Providers
[EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
Gang Chen wrote: On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote: We are using Level 3. At this point, changing carrier is not an option. Gary, I use Level(3) with G729a and RFC2833. No problems, no requirement for inband G729. -- Kristian Kielhofner I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729. It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto with g729, I got warning saying something like * does not support inband for g729 and sutomaticlly switch to rfc2833. If I set dtmf=g729, there is no warning but I have the same problem. This tells me that Level3 does use inband for g729 or maybe I am doing something wrong . Gary Gary, I'll restate what Kristian just said above. You do NOT need inband for Level 3. Set dtmf=RFC2833. Do you have the correct g729 codec licenses installed? This may be more of a transcoding issue than anything else. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
Joshua Colp wrote: Andres wrote: I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729. It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto with g729, I got warning saying something like * does not support inband for g729 and sutomaticlly switch to rfc2833. If I set dtmf=g729, there is no warning but I have the same problem. This tells me that Level3 does use inband for g729 or maybe I am doing something wrong . You are doing something wrong. Nobody uses inband DTMF for G729 because it does not work. Do a sip debug to make sure your sip.conf entry with the dtmf=rfc2833 is being used. I'll chime in since nobody has yet corrected this... it's dtmfmode=rfc2833 not dtmf=rfc2833 Yeah that's what I meant. Total lack of caffeine this morning. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving Asterisk's DNS support
Kristian Kielhofner wrote: On 6/20/07, Steven [EMAIL PROTECTED] wrote: I could understand if it couldn't register to an ITSP or similar. But, (I had this happen today) asterisk takes forever to start up and SIP phones can not register to it. DNS should not need to be used for anything in asterisk except registering to VOIP providers and maybe external SQL from the dialplan. If there are reverse lookups being done, I do not see the output of it. Steven, If you are using a hostname for an ITSP and DNS fails, it will take FOREVER for the SIP channel driver to load/reload/do anything that requires a DNS lookup. This will in some cases block the rest of Asterisk but will certainly make anything that depends on SIP break - until the DNS request finally fails. I have started a new thread on -dev about this... I experienced this exact problem last night on my personal box. My sip provider went un-reachable (Teliax requires the use of hostnames). When that happened, I couldn't even call my local phone extensions. Everything SIP was locked hard until it finally timed out. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
