Re: [asterisk-users] Asterisk and Packetcable
I am also interested in this. Sent from my Verizon Wireless BlackBerry -Original Message- From: Carlos Alberto Bernat Orozco [EMAIL PROTECTED] Date: Mon, 10 Mar 2008 11:55:25 To:asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Packetcable Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat ___ -- 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 ___ -- 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] T1 Timing Troubleshooting
Jon, did you ever discover a solution to your problem. I'm in the same boat. On Sun, Dec 2, 2007 at 1:22 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM calls (when it comes to faxing) must be having timing issues since they intermittently fail. I found what seems to be an issue in zaptel.conf (timing source for the Comdial side was 2 - changed to 0), but I don't know if that's it. I've also turned off echo cancellation. Any other thoughts on why I may be having what seem to be timing issues? Also, is timing passed through on dynamic spans bridged calls? And is there a way to verify this? Thanks! - /etc/zaptel.conf (16 channels on each PRI): loadzone=us defaultzone=us #Sangoma A400 [slot:7 bus:1 span:1] fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxoks=11 fxoks=12 dynamic=eth,eth1/00:50:c2:65:d0:3c/0,24,1 dynamic=eth,eth1/00:50:c2:65:d0:3c/1,24,0 # bchan=25-47 bchan=25-40 dchan=48 # bchan=49-71 bchan=49-64 dchan=72 - /etc/asterisk/zapata.conf: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes ; Turned echo cancellation off 11-15-2007 due to possible fax issues on bridged calls. echocancel=no faxdetect=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no overlapdial=yes ;Sangoma A400 [slot:7 bus:1 span:1] context=from-zaptel group=0 signalling = fxs_ks channel = 1-8 context=from-internal group=1 signalling = fxo_ks channel = 11-12 ; First port on foneBRIDGE2 - This is the PSTN side group=2 signalling = pri_cpe context=from-pstn ;channel = 25-47 (for a full PRI) ; Channels 25-40 are for a partial PRI (16 channels) channel = 25-40 ; Second port on foneBRIDGE2 - This is the Comdial side group=3 context=from-comdial signalling = pri_net ;channel = 49-71 (for a full PRI) ; Channels 49-64 are for a partial PRI (16 channels) channel = 49-64 -Jon ___ --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 ___ -- 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] Disappearing B-Channels
Tilghman, Could you clarify what you mean when you say you added usleep(1) to the end of the manager thread? I do not have enough experience to follow what you're saying. Are you talking about adding this command to the end of the manager.conffile? - Mark ___ -- 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] Disappearing B-Channels
OK thanks for the effort. What's a way to look for IRQ misses in linux? On Feb 20, 2008 12:33 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 19 February 2008 22:20:06 Mark Greene wrote: Could you clarify what you mean when you say you added usleep(1) to the end of the manager thread? I do not have enough experience to follow what you're saying. I added the comment about what I did to fix it in the source, for those who can follow. If you can't follow, then don't worry about it. Are you talking about adding this command to the end of the manager.conffile? No. -- Tilghman ___ -- 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 ___ -- 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] Disappearing B-Channels
I could do that. The only issue is that I don't understand why others with my setup have not had to do the same. What's unique about my TDMoE setup that makes it intolerant to channel restarts? I did everything by the book. On Feb 11, 2008 11:06 AM, Jared Smith [EMAIL PROTECTED] wrote: On Sun, 2008-02-10 at 22:01 -0600, Mark Greene wrote: Let's compare. Right now I am restarting asterisk when convenient every hour, and that's keeping the symptoms at bay. Have you tried setting resetinterval=never in zapata.conf? By default, Asterisk resets all *idle* bearer channels (B-channels) every hour. (When you say you're restarting every hour, that gave me the idea that maybe the B-channel restarts have something to do with it.) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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 ___ -- 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] Disappearing B-Channels
I don't think it's my telco, I think it's my TDMoE setup. Does that sound possible? I've never had problems with the circuit until I moved it from a standard digium PRI card to a TDMoE device. Also, if I restart asterisk, all the b-channels come back. Thoughts? On Feb 10, 2008 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 10 February 2008 01:44:38 Mark Greene wrote: In my efforts to solve a mystery of asterisk slowly loosing it's ability to take incoming and outgoing calls I set asterisk to restart b-channels every 60 seconds hoping I would find something odd after some time. So now I am looking at the CLI a few hours later and look what happens when asterisk restarts the 23 b-channels I have. pbx1*CLI -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 == Primary D-Channel on span 1 down [Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame while link down == Primary D-Channel on span 1 up -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 pbx1*CLI That's the output while I've been writing this email. Those are TWO restarts of the b-channels. Notice I am missing a seizable amount of my 23 b-channels. Where are they going?! How do I find out? I've recompiled my asterisk, zaptel, and libpri to the most recent versions but that's made no difference. You probably have noise on your T1 circuit, which is causing the PRI signalling to become corrupt. If this continues, expect that the T1 circuit will go down from time to time, for a few seconds each time. Your solution is to call your telco and ask for a loopback test. -- Tilghman ___ -- 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 ___ -- 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] Disappearing B-Channels
Kevin, Let's compare. Right now I am restarting asterisk when convenient every hour, and that's keeping the symptoms at bay. However if I let the problem get to the point that it won't take ANY incoming / outgoing calls then I cannot even initiate a restart from the CLI. I have to back all the way out of the CLI to the system prompt and actually call the init script to restart the asterisk service itself. Is this similar for you? What is your system? I am running Dell Poweredge P4 2.8 Ghz 512 MB RAM 80 GB HDD on Software RAID 1 CentOS 5 Kernel 2.6.18-53.1.4.el5 i686 Asterisk 1.4.18 Zaptel 1.4.8 Libpri 1.4.3 asterisk-addons 1.4.5 On Feb 10, 2008 5:06 PM, Kevin Kiely [EMAIL PROTECTED] wrote: Mark, I thought I would also mention that I am still having similar issues even after updating to the latest Asterisk, Zaptel and Libpri. Although I am using a Sangoma, we have similar symptoms with a restart fixing it. I am starting to wonder if I must go back to a Digium card. We originally switched away from the Digium TE110 because of interrupt issues, I think that the interrupt issue has been remedied now. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mark Greene *Sent:* Sunday, February 10, 2008 11:15 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Disappearing B-Channels I don't think it's my telco, I think it's my TDMoE setup. Does that sound possible? I've never had problems with the circuit until I moved it from a standard digium PRI card to a TDMoE device. Also, if I restart asterisk, all the b-channels come back. Thoughts? On Feb 10, 2008 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 10 February 2008 01:44:38 Mark Greene wrote: In my efforts to solve a mystery of asterisk slowly loosing it's ability to take incoming and outgoing calls I set asterisk to restart b-channels every 60 seconds hoping I would find something odd after some time. So now I am looking at the CLI a few hours later and look what happens when asterisk restarts the 23 b-channels I have. pbx1*CLI -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 == Primary D-Channel on span 1 down [Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame while link down == Primary D-Channel on span 1 up -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 pbx1*CLI That's the output while I've been writing this email. Those are TWO restarts of the b-channels. Notice I am missing a seizable amount of my 23 b-channels. Where are they going?! How do I find out? I've recompiled my asterisk, zaptel, and libpri to the most recent versions but that's made no difference. You probably have noise on your T1 circuit, which is causing the PRI signalling to become corrupt. If this continues, expect that the T1 circuit will go down from time to time, for a few seconds each time. Your solution is to call your telco and ask for a loopback test. -- Tilghman ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.0/1268 - Release Date: 2/9/2008 11:54 AM ___ -- 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 ___ -- 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
[asterisk-users] Disappearing B-Channels
In my efforts to solve a mystery of asterisk slowly loosing it's ability to take incoming and outgoing calls I set asterisk to restart b-channels every 60 seconds hoping I would find something odd after some time. So now I am looking at the CLI a few hours later and look what happens when asterisk restarts the 23 b-channels I have. pbx1*CLI -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 == Primary D-Channel on span 1 down [Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame while link down == Primary D-Channel on span 1 up -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 pbx1*CLI That's the output while I've been writing this email. Those are TWO restarts of the b-channels. Notice I am missing a seizable amount of my 23 b-channels. Where are they going?! How do I find out? I've recompiled my asterisk, zaptel, and libpri to the most recent versions but that's made no difference. - Mark ___ -- 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 mem leak behavior?
Kevin, After upgrading to the latest build of everything have you seen the problem anymore? What's your hardware and software configs? Maybe we can find a similarity in our systems. - Mark On Jan 29, 2008 9:53 AM, Kevin Kiely [EMAIL PROTECTED] wrote: Mark, I thought I would chime in here on your problem. Oddly, I have having the same issue with a PRI with similar symptoms. The odd part is that I have never had an issue like this with a asterisk PRI setup. My setup is a PRI with a Sangoma card with the exact same issue with 1.4.14. After a few days we are unable to communicate with the PRI, The D-channel goes offline as well but the physical circuit stays up with no alarms. It doesn't give one a comfort level with uptime. I had also re-compiled the asterisk 1.4.14 along with zaptel and libpri sources and it still failed. I have since updated to the latest asterisk, zaptel and libpri .17 with the hopes that it will be fixed. I thought perhaps the card may have had an issue but now I am beginning to wonder. Kevin -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mark Greene *Sent:* Tuesday, January 29, 2008 9:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk mem leak behavior? I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date: 1/28/2008 9:32 PM ___ -- 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 ___ -- 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 mem leak behavior?
I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark ___ -- 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
[asterisk-users] Asterisk mem leak behavior?
So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 I'm using TDMoE with my PRI using a product called fonebridge from a company called redfone. They require that I use their own build of zaptel and I am trying to figure out if the problem is with them or something else. The TDMoE traffic is running over a dedicated interface on the asterisk server. Nothing but TDMoE traffic goes over this interface, it does not even have an IP assigned to it. After a about 6-7 days asterisk will stop passing calls through the PRI. It will continue to accept calls from extension to extension or to voicemail, etc. But nothing over zap channels. From the CLI if I check the status of the channels it shows all is well. If I use zttool it shows all OKs. And the lights on the fonebridge itself are green across the board as well. If I issue the command from the CLI to restart asterisk restart now it drops me to another CLI prompt as if it ran the command, but it didn't asterisk continues to run. It takes me dropping out of the CLI and issueing an init command to restart /etc/init.d/asterisk restart. Once asterisk restarts all is well and things keep moving, but 5 or so days later the same thing happens. What's the best way to trouble shoot this? Any and all comments are welcome. Let me know if you need more details. - Mark ___ -- 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
[asterisk-users] MySQL + Realtime + SIP Registration
I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using the realtime load command. This is after i added sipusers = mysql,asterisk,sip_users to my extconfig.conf file. However I don't know what to do to get asterisk to look at that table when a request to register comes from a sip peer. I understand that sipusers and sip peer are contradictory but they are all defined as peers. ___ --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 + Hotel Management System
I am interested in doing some hotel installs of asterisk, but I too am confronted wtih the hurdle of making it play nice with reservation systems such as Micros System's. Have you made and discoveries? - Mark ___ --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
Re: [asterisk-users] Micros-Fidelio - billing in hotel
Have you made any progress. I am interested in the same thing. - Mark ___ --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
[asterisk-users] delete=yes is not working
Hey guys. This is the setup that I have for a voicemail account. 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=yes It emails me the voicemail, but it does not delete it from the system afterwards. I have also tried 1509 = 1509,Mark Greene,[EMAIL PROTECTED],attach=yes|delete=1 Any ideas on this? - Mark ___ --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
Re: [asterisk-users] delete=yes is not working
I have googled and I do not understand how the pager field is what is causing the problem. Could you explain? ___ --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
Re: [asterisk-users] delete=yes is not working
OK that makes a lot more sense. Thanks for the explanation. - Mark ___ --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
[asterisk-users] HowTO configure voice T1
Alright guys here is my question. What is do I need to set switchtype, and signalling to in zapata for a voice T1. This is not a PRI. I cannot say that enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice channels. There is not a D Channel. It runs from one office to another and USED to plug into two opt. 11c but now one end is going to plug into an asterisk box. - Mark ___ --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
[asterisk-users] Voicemail to email
Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark ___ --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
Re: [asterisk-users] Voicemail to email
In my case it was never any confusion over what needs to be configured in asterisk. I was wondering what mail program asterisk used and what needed to be configured with it. In my case I had to set up sendmail on my system to relay through our internal mail server. sendmail.mc was the file I had to modify and SmartHost was what I had to append to. ___ --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
[asterisk-users] ARI help
I am trying to use ARI for call monitoring. Recording conversations and such. The problem is that I don't use AMP, and don't have any sort of a database for CDR setup. It is all stored in the CSV file by default. When I setup ARI I tell it to go into standalone mode, and I set the asterisk manager username and password that was defined in manager.conf, but it also wants a cdr username and password that I don't know exists. Also, EVERYTIME I leave the callmonitor module active, it tells me that it could not find the DB extension and to check AMP, asterisk, and main.conf. So do I NEED to have AMP installed for call monitor to work? How can I setup ARI with JUST ARI and a STANDARD asterisk install. No AMP or SQL. - Mark ___ --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
[asterisk-users] Best Hardware for Asterisk Server?
Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. - Mark ___ --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
Re: [asterisk-users] Best Hardware for Asterisk Server?
I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. ___ --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
Re: [asterisk-users] chan_oh323 early media
I am a a little confused on how to get h323 working on asterisk. Could you please point me towards specific resources you used? voip-info.org seems to keep me in a loop of info. ___ --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
Re: [asterisk-users] Best Hardware for Asterisk Server?
Wow Doug thanks for the specs. This has really helped. ___ --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
[asterisk-users] Asterisk and MiniITX setups
How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark ___ --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
Re: [asterisk-users] Asterisk and MiniITX setups
Doug, This is a great price but I have a couple questions, as I am not very savy with these systems. When you say, replace the ROM chip, what do you mean by that? Replace it with another ROM chip, and how would I go about learning to program one, etc. As for the PCI riser card, I do not see that there would be room or even a slot for a pci card in the system you linked. to. My last and most important question is, can an 800 mhz chip handle much of a load at all for asterisk once you put a PRI or FXO/S card in? I would love to pursue this. Thanks, - Mark ___ --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
Re: [asterisk-users] Asterisk and MiniITX setups
I will be doing transcoding though. ___ --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
Re: [asterisk-users] Asterisk and MiniITX setups
Nathan, what hardware are you running? ___ --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
[asterisk-users] Avaya to Asterisk via H323
I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and where do I need to look for setting it up on centOS 4.4? Thanks in advance, - Mark ___ --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
Re: [asterisk-users] Avaya to Asterisk via H323
Andrew, I am not so reluctant when it comes to figuring out how to link the two systems once I have asterisk working with h.323. The email was asking if someone could point me in the right direction of how to setup h.323 on asterisk. I am confident that I can handle the config from there. Do you have any thoughts on that? - Mark ___ --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
[asterisk-users] Searching the list
Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --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
Re: [asterisk-users] Searching the list
thanks for the tips. ___ --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
