[asterisk-users] How to set name of call wav recording file in outgoing/call file?
When I mv a file to /var/spool/asterisk/outgoing in order to place a call from a user extension that will always be recorded, what parameter do I set in the call file in order to specify an exact name for the wav file? This is on Trixbox and at the moment I'm considering setting an extra variable and calling through a new calling context that reads this variable and sets the recording name. -HJC ___ -- 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] End to end call monitoring?
We're having some difficulty tying together our Cisco and Audiocodes syslogs with our Trixbox asterisk logs. We'd like to have some way to split out a single call from all the activity going on at one moment. Obviously NTP is the first step for this, but we haven't found any means to tie the logs from the Cisco (which if we're lucky gives a UDP port number) and the Audiocodes (which rarely tell which of the 24 phone lines is generating the error or warning message). -HJC ___ -- 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] does the meetme module still require an external timing source?
On Tue, Apr 1, 2008 at 11:03 AM, Mike Trest - Personal [EMAIL PROTECTED] wrote: At 01:13 PM 4/1/2008, you wrote: Is app_conference stable now? I've never made it through a thousand calls without a crash. (With a busy call center this doesn't take all that long.) I have deployed a MEETME conference bridge based on a FARM of asterisks with 6,000 conference ports active using basic meetme() with a very complex IVR front end that we wrote in perl for the customer specific needs. Still in use after 3+ years. Hundreds of thousands of total participants and millions of minutes later, still running. Very happy with results. The only problems I have with MeetMe is that the monitoring controls don't work very well, a single channel can derail a conference and tie up the server, it eats CPU and adds latency. (Oh yeah, and there is that ZAP clone card timer chip requirement.) Other than that it's okay. I'd love to have a solid alternative in Asterisk, but I haven't see it yet. -HJC ___ -- 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] breaking into asterisk channel
On Tue, Apr 1, 2008 at 7:29 AM, Chaya Zipora Rosenberg [EMAIL PROTECTED] wrote: Hello, I am setting-up a system to place outgoing calls for a certain number of minutes (as allowed per the customer's account). I would like to break into the long distance channel to announce 1 minute left, etc. What asterisk command can I use to do this? MeetMe. You will be blending in multiple inputs into a single output channel after all. Are you going to tell both sides this, or just one? -HJC ___ -- 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] does the meetme module still require an external timing source?
On Wed, Mar 12, 2008 at 1:57 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 16:27, Wed 12 Mar 08, Steve Totaro wrote: Try Callweaver. Thanks, Steve Totaro or app_conference for asterisk. That does the trick for me on OpenBSD where you dont have ztdummy. Is app_conference stable now? I've never made it through a thousand calls without a crash. (With a busy call center this doesn't take all that long.) -HJC ___ -- 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 But I Would Rather See People Running Asterisk on a Real Server than an Emachine
Out of stock now. Any war stories about running Asterisk on a serious blade setup? Will you ever hire Wesley Snipes to flog them at a convention? -HJC ___ -- 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, HP, Digium, homebrew - what do you use
Local (Indian) vendor, Intel(R) Pentium(R) 4 CPU 2.66GHz, 1 GB RAM, Two 80GB IDE disks, X100P clone. Supports 15 agents as SIP/ulaw to IAX/G.729 bridge running Vicidial ___ --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] Crontab script to check health of Asterisk server?
On 7/16/07, Dovid B [EMAIL PROTECTED] wrote: Define health. I was working on but gave up on it (no time) to have serverA call serverB. ServerB has an agi that it runs that stores info in DB. if serverB doesn't get a call then we know that there are issues (and run the script vice versa). Yeah, have machine A call machine B every five minutes and timeout the dial after the CID is delivered but before the normal IVR on the DID picks up. (Perhaps this needs a dedicated DID, but that's only $7 a month.) I suppose our FreePBX server would be the best target for this. -HJC ___ --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] Question about dnsmgr
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to move existing connections when you see a DNS change or do you leave the existing connections the fnord alone on their current IP addresses and simply use the DNS change for new connections? -HJC ___ --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] X100P Clone
On 6/6/07, John Novack [EMAIL PROTECTED] wrote: Henry Cobb wrote: Why would anybody plug a telephone line into an X100P clone? ??? What else would one plug into it? We just use them as clock cards for MeetMe and trunking. -HJC ___ --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 Codec
On 6/7/07, Ricardo Martins [EMAIL PROTECTED] wrote: We use G.729. Consumes only 35kbps of bandwidth and has a level 4 (from 0 to 5) of voice quality. We still have very poor public data networks here in Brazil that makes G.711 a very high bandwith consunption codec for us. 35kbps sounds very large. We only use 20 kbps untrunked and 13-15 kbps when using IAX trunks. Have you verified this bandwidth usage? -HJC ___ --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 call quality detection
On 6/6/07, carl Lougher [EMAIL PROTECTED] wrote: Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? I chart VNAKs per hour. -HJC ___ --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] X100P Clone
On 6/5/07, Jared Smith [EMAIL PROTECTED] wrote: Most of the clone cards don't support far-end disconnect supervision, so you'll have problems where Asterisk can't tell that the other party has hung up the phone. You'd be better off to buy a modern Asterisk telephony card. Why would anybody plug a telephone line into an X100P clone? And when will Digium offer affordable one-port cards again? -HJC ___ --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 call quality detection
On 6/6/07, Matt [EMAIL PROTECTED] wrote: I chart VNAKs per hour. Would you care to share how you accomplish this? What programs do you use? grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | uniq -c Needs a bit of an adjustment between the 1-9th and 10th-31st of the month so I'm looking for something to chomp this automatically. -HJC ___ --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] X100P Clone
On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to buy a X100P clone and would like some feedback about this card. Does anyone already used this card? Does anyone recommend it ? or not? I'd appreciate any comments. If you have a new 3.3v only motherboard then make very sure that the brand that you buy supports this or your system will refuse to boot with the card inserted. A lot of X100P clone cards have the 3.3v notch in their PCI interface, but do not support 3.3v operation. -HJC ___ --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] NAT
On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi All!! I have my asterisk working in my house (working with mandriva 2007 and asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the way of making work my asterisk in a real enviroment. Seems that the problem of NAT is a big problem. How can I sort out this, a mean crossing the NAT and having asterisk connected? If you want to receive calls and not just place them and you have a broadband connection with a dynamic IP then your server must register with the VoIP provider and I suggest using IAX with the proper UDP port assigned to your Atrisk server. -HJC ___ --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] Additional commands for MeetMeAdmin
Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC ___ --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] polycom random reboots
On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? Our Polycom 3s and 5s ship with flaky power supplies and tend to reboot all of the time (especially in India...), so we found replacement non-Polycom power supplies and they are much more stable. -HJC ___ --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] Which parameters of a live Asterisk server would you monitor ?
On 3/20/07, Olivier [EMAIL PROTECTED] wrote: Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? The tools I tend to use are vmstat, iftop (all VoIP, all the time), show registry and df. -HJC ___ --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] voip-info.org status update
On 3/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Obviously you didn't read Google's research paper on drive failures. This one? http://labs.google.com/papers/disk_failures.html -HJC ___ --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] Newbie Question
On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? Setup the demo IVR on your Atrisk box and call that from your xlite softphone. The entire call will be on your local network so you'll be able to see if the problem is local or not. -HJC ___ --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] How many outgoing phone line/voip account do I need?
On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote: On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote: More importantly, how many calls per day and how long per call. Then you can figure out the other bits. He wants to make 50 simultaneous calls. What difference does the length and frequency make. His vindictive dialer isn't playing while it is listening to rings or busy signals. So there is an impact on CPU usage from the length of time it takes the average victim to hang up. -HJC ___ --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] How many outgoing phone line/voip account do I need?
On 3/13/07, Chris Bagnall [EMAIL PROTECTED] wrote: His vindictive dialer isn't playing while it is listening to rings or busy signals. Forgive my ignorance, but what on earth's a vindictive dialer? Is it one with a strong sense of revenge? :-) A normal predictive dialer determines from agent behavior when will be the most convenient time to deliver the next call to them. A vindictive dialer uses arcane arts to determine the least convenient time to deliver the call to the target. Is it when they are about to sit down for dinner, when they are about to step out or when they are taking a bath? Many factors have to be adjusted to maximize the inconvenience of the call. The dinosaur telephone companies are the main users, but the free vacation seminar companies are stepping up their deployments. Personally I never answer calls from area code 666 anymore. ;-) -HJC ___ --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] Which VoIP router and switch to use for medium size business
On 3/10/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/10/07, Henry Cobb [EMAIL PROTECTED] wrote: So get a second broadband connection and run only voice on it. Has anyone tried this? I have been thinking about this. We're getting so much spam that I think it's taking up too much of our bandwidth. I'm wondering how much bandwidth all the script kiddies take up scanning things as well. That won't be a problem if you've got almost every port blocked at the firewall. Sell this to your client on the basis of uptime. You wouldn't want your phones to be unusable just because your ISP has a routing problem. Buy two links of the same size from two different kinds of providers and put the tiny trickle of voice on the best link and your hordes of data on the non so good link. Then sign up for two different VoIP providers and use whichever is best on your best internet link as your primary with the other as backup. Then all you have to do is ensure that everything on the phone to internet route has UPS protection. (A standby PBX PC wouldn't hurt either.) -HJC ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
It looks like somebody has fixed the X100P v3.3 problem in an outbreak of sanity. (Pity that Digium seems to be in such a hurry to not take my money. Well, other than the G.729 tax.) http://cgi.ebay.com/Low-Profile-Authentic-X100P-SE-FXO-Digium-Asterisk-VoIP_W0QQitemZ130088348688QQcategoryZ99269QQssPageNameZWDVWQQrdZ1QQcmdZViewItem?hash=item130088348688 Anybody tried this in a 3.3v only server yet? -HJC ___ --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] Which VoIP router and switch to use for medium size business
On 3/9/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I am thinking of going with HWEC and also using a good QoS switch. Right now there is only one switch (don't remember the name) and it is handling all the VoIP and data traffic. Sometimes voice breaks, and it must be because of interference from data traffic. But this is not a very serious problem and one switch with QoS should be able to handle it. Am I right here? Even if someone starts using P2P software. Not a chance. Most of your traffic is doubtlessly downloads such as web and spam mail. A QoS device on your side of the link can only reduce the amount of stuff you send (which isn't much) and can't do anything about the massive amount of downloaded stuff that is getting in the way of voice packets being sent to you. So get a second broadband connection and run only voice on it. -HJC ___ --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] Newbie Question
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote: [test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? I use it on Linux and it does. -HJC ___ --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] Empty Wildcard TDM400P as a MeetMe timer.
I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC ___ --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
