Re: [asterisk-users] asteriskt38.com
only condition would be that you do not use it for a commercial use, i.e. you don't try to sell a t.38 module for asterisk. If you want to retain any control of what it is used for, you better re-register it. Once it expires and some one else gets it, you have no say in the matter. John ___ -- 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] Amazing show uptime
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. That would be 1980, right? Only if this was 2018... No. Looks like some signed/unsigned int error somewhere. The lost bios makes more sense than this. If this were the case don't you think we would have a LOT more bug reports? John ___ -- 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] IVR response of the pound key
But each time the pound key is pressed, file dir-intro.gsm will always be played. Can any one tell me the reason? Because you have enabled the directory in your IVR. Uncheck the Enable Directory box on the IVR screen and you can then use the octothorpe. By the way,if I want to modify the build-in IVR, which configuration file should be edited? Thanks in advance. Your best bet is extensions_custom.conf. This way you can define your own without the hassle of having to redo it every time you upgrade. John ___ -- 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] Fwd: USA Lata AreaCode Database
Why everybody charge so much for this information, why this information could not be free ? Probably because Telecordia makes a decent amount of money from selling it. If they gave it away that revenue stream would dry up. John ___ -- 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] Implementing an Asterisk Server behind a MeridianNorstar
Does anyone make an interface card that can integrate with the digital input of the Meridian. Not the optimal solution, but it allows for the current infrastructure to be retained. By digital input do you mean a T1 interface? If so then yes several T1 interfaces are available. However I think you mean is there a gateway to use the Meridian/Norstar phones with Asterisk. If so, yes there is a company that makes a gateway to use the Nortel p-phones with a SIP based system. However past experience has shown that for the less than the cost of the gateway, I could replace the phones with IP phones and eliminate another point of failure and the hassle of configuring it. John ___ -- 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] Implementing an Asterisk Server behinda MeridianNorstar
Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the Joseph, Now I'm pretty sure we are not talking about the same things. Let me see if I have the correct picture in my head. I now think you have a Norstar in one office and an asterisk system in another office and want to allow them to send calls between them. Is this correct? Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? The new Polycom 670 has a gig interface but at this point I'm not sure why you need that. Are you thinking that if the Norstar phones and lines can't be used, that you would need the phone to have a switch to share the Ethernet connection? Sorry for the confusion but I just want to make sure I know what you need before making a recommendation. John ___ -- 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] Diagnosing dropped calls...
-Original Message- Subject: [asterisk-users] Diagnosing dropped calls... I have a system that is driving me nuts. My customer is running Asterisk 1.4.20.1 on a CentOS 5.2 server. It is a purely SIP and IAX2 service with no cards installed and it uses ztdummy from Zaptel 1.4.11. They use Teliax for calls to the USA and Protel for calls in Mexico. The problem is that users complain that their calls get dropped. Sometimes every few minutes and sometimes after a very long call over two hours so there is no clear pattern. There is a I just went through this same scenario. The customer originally using Teliax over ATT DSL. Incoming calls would sometimes have a delay of 1-2 seconds before connecting and some would just drop at random intervals. We would generally see in the logs that our connection to Teliax would often show lagged or unresponsive around the time of the drops. Testing with ATT revealed that the customer was 18,034 feet from the CO so we decided to move them to a cable modem from Charter. This was done as it was the only other internet provider short of a dedicated T1. Things improved but did not completely subside. The delay on answering was becoming the major complaint. We decided to try a different provider so an account with Junction Networks was added. This eliminated the delay immediately but did not eliminate the dropped calls. One of the techs at Teliax told us that the delay was an issue caused by Asterisk 1.4 trying to talk with Asterisk 1.4 server. We tried their work around but it did not eliminate the delay issue though it did reduce the issue. We also had Charter monitor the service for nearly two weeks in which time they replaced the modem 4 times and claimed to have repaired connectors, replaced repeater nodes and changed other connections to no avail. The customer lives and dies by their telephone so we moved them to Cbeyond SIP Connect. All of the issues are now gone. The is no delay, they some times spend hours at a time on training calls with no issues. I also did a bit of testing with the Junction Network account after moving to Cbeyond. This was done since we still had money in the account anyway. Again no more issues with dropped calls. The key is in an Internet connection that is either very clean or supports QoS. The Cbeyond connection seems to have both. Obviously we don't have QoS back to Junction but with the cleaner connection it doesn't seem to matter. Also if you trying to use a cable modem, make sure your using a business class service and not a residential service. Also use mtr (mytraceroute) to look at the latency between your server and the proxy. Under Charter and ATT, both of which were business class services, the mtr would sometimes show up to 20% packet loss at various systems between us and the proxy. Also try different proxies. For us the best one was typically either Atlanta or Denver. Keep in mind that I'm not bashing Teliax or Junction. I use both Teliax at home and Junction in our office, over a cable modem provided by Time Warner and the only issue I have is the occasional delay on answer with Teliax. However that one is the home line and the delay seems to work great for telemarketers. John ___ -- 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] Diagnosing dropped calls...
Try dropping the IAX2 and only use SIP. Don't ask why? Well in our case we were NOT using IAX at all. Strictly SIP. You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. So true, hence the reason I suggested using mtr to check it. Many times in our case we saw gateways between networks that were dropping packets presumably due to overload conditions. RTP traffic over UDP would add far more load than the ICMP packets used for mtr. Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly (point to point) with G729 or just get a real ISDN or POTS lines. With this particular customer we initially had unrealistic time and budget constraints. We had 11 days between when they contacted us and when the system had to be in place. With an average install time around here of 30-35 days that ruled out the T1. Their initial monthly telephone budget was $150 per month. Like I said, unrealistic. Usually, once they perceive a problem, then even if the other side of the call is on a cell and the cell drops the call, Again so true. This customer got to the point that if a call dropped when they we on their cell phone, regardless of the fact that the call was not going through our system, we got a trouble call. Thankfully, a week after cutting over to the T1 we called them to make sure they hadn't found someone else. 8) John ___ -- 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] Minimum upload speed for Asterisk?
to suffer. IVR prompts and MOH frequently have slight pauses from the outside, but sound fine from inside calls. If the pauses your talking about are at the beginning of the call but are fine after the call is answered, I'd suspect latency issues. If the pauses are no longer than just glitches, static or pops then suspect jitter. If the pauses are longer, and affect IVR, MOH and voice traffic equally, then look for packet loss. If the voice traffic is fine but the IVR and MOH are affected, look for I/O bottlenecks in your hardware. Try using My Traceroute (mtr) to your SIP provider. This may point out packet loss and latency issues. I say may because ICMP traffic could be handled differently than SIP/RTP traffic. Is 384kB up too slow? Depending on the codec used, this should be sufficient for 4 or more simultaneous calls as long as the latency, packet loss and jitter don't kill it. John Faubion ___ -- 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] Paging for analoge devices
need auto answer or how else would the audio come out of the speakers? One of the lowest cost solutions I've used for this is a Grandstream BT-200. You can buy the phone for around $50, it supports auto answer, and it has a miniature audio jack already on it. Just run the output to an amplifier and speakers and your good to go. John ___ -- 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] ISPBX Announces COGOBLUE Interface andPBX Appliances
As far as a license is concerned, we do not ship with any codecs that require licensing (we support them) and when someone purchases an ISPBX PBX system, the license for using What Kristian was asking is, what license does the software you have written use? Is it GPL? Seeing that many of us only run open source software due to being burned by proprietary software and systems, Cogoblue would be better received if it were open source. John ___ -- 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] More Broadvoice woes. Who's fault could thisbe?
Forwarding isn't on. If I call from my cell: fax machine. Call from a land line: call gets through. Call from Skype: fax machine again! Ok now that is bizarre. Three different sources and two different destinations. Looks like the only common point would be BV. Sounds like your going to get to spend some time on the phone with BV. If only phone numbers were trace routable like IP addressed, to see where the heck my calls are going. I'd really like to see that feature as well! John ___ -- 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] More Broadvoice woes. Who's fault could this be?
cell phone. When I do, I get a fax macine. Debugging SIP shows NO call activity what so ever. Make sure you don't have it forwarded to another number at BV. John ___ -- 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
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? Thanks, John ___ -- 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] Is Asterisk ready for Prime-Time?
I reboot every evening :) Drew, what's the uptime on your asterisk process on that box that's been up for 193 days? I too restart the asterisk process every night as part of the cron process. Many people here seem to be under the impression that restarting the application every day is a bad thing. Having worked with carrier grade systems for 20+ years, I can tell you that even these systems restart the application during the days slow period. Granted these are usually two separate systems for redundancy but the typical method is to: 1) Unsync the two systems 2) Run system testing on the inactive side 3) Restart the inactive side 4) Resync the data between the systems 5) Switch the active and inactive processors 6) Repeat steps 1-4 on the newly inactive side Now don't think that smaller PBX or key systems are all that different. I know that the Meridian systems go through a similar process each day. On the SL1 systems there is a garbage daemon that runs every day. This daemon restarts the application to clean up RAM allocation. The Norstar key systems do this as well although the reset only takes about 2 seconds. Since everything is stored in flash memory, it is a quick way to make sure any glitches in RAM are cleaned up. Interestingly, the restart of Asterisk on my system only takes 3-4 seconds. Actual call processing is probably only affected for less that 2 seconds. Done during our night time activities no one ever notices. I've had some argue that a restart shouldn't be done because of the possibility that the system might not come back up. While this is potentially true, it will be because a file was changed without restarting or reloading asterisk. Yes this can happen though the likelihood is very small. At least it should be on a production system. One other thing to point out, if you are the type to constantly upgrade to the latest and greatest, you can expect to have issues. Once you get the system on a stable setup, the only reason for upgrading is if the new version fixes some problem that you have. Again some argue that security vulnerabilities would require the upgrade but that isn't always the case. If your system is a closed network, for example, your connection to the outside world is strictly analog and your network isn't shared with your computers, none of the security concerns would every matter. Now think back to that key system you revered for just working, did it have any outside connections that a hacker could exploit? Not likely. I have one system that we installed nearly a year ago. The only time it has been down was due construction workers cutting the main power feed to the building, between the building and the generator. It took them 10 hours to fix it and the UPS lasted over 4 hours. That was 200 days ago however asterisk was restarted about 8 minutes after midnight. As they say, your mileage may vary, but I don't think restarting asterisk is a bad thing. John ___ -- 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] Is Asterisk ready for Prime-Time?
Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? I actually work as a software engineer for a big telecom manufacturer to remain unnamed. I use Asterisk at home and I built a system for my aunt's real estate office mainly because she was quoted $83K+ over 5 years for a 12 station Toshiba key system. I now get calls to build more of them mainly for real estate offices that have seen other systems I have built. I probably should become a full blown reseller but I don't see me making enough money to walk away from my daytime gig anytime soon. On second thought, I guess if I were charging $83 large per customer maybe I could! John ___ -- 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] Is Asterisk ready for Prime-Time?
For such a small system there is no earthly reason for it to be 10 percent of that, even on a 5 year lease. I know that EVERYTHING is big in Texas, but that is nothing more than highway robbery. I fully agreed, that's why we built her an Asterisk based system. Splitting this up they wanted $724 per month for the hardware and maintenance. This did include a special kind of lease where they could upgrade as necessary even if it required them to change out the system to do the upgrade. I'm not sure what that is worth but I'm fairly sure it shouldn't cost this much. The monthly contract for the Integrated PRI was another $675 per month. My aunt couldn't see how she was going to afford that so she called me for advice. I originally steered her toward a key system until I realized she would eventually need 35-40 stations. So we rolled our own asterisk based system. John ___ -- 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
when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Remember that the service providers such as ATT, Cingular, Sprint, Verizon and so forth, subsidize the cost of the phones because they make it up over the course of the contract. Hence the reason that some phones that have an initial cost when sold with a 1 year contract may be free initially with a 2 year contract. Even some VoIP phones and ATA's are done this way but only through service providers. Take the subsidies away and that iPhone is pretty pricey. John ___ -- 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] NIN Ghosts music (free download) safe for MOH?
But, just to clarify, please remember that using music as MoH is considered a public performance, and if the pieces in question do not include a buyout license *for the performance Ok now I am curious, if a radio is playing in a store, a restaurant or at the beach, wouldn't that be considered a public performance? And even though the radio station has already paid the license fee, does this mean that the person who owns the radio is also subject to these fees? I know of several key systems with FM radio cards providing MoH and I've often wondered about the ramifications of that setup and the music industry. John Faubion ___ -- 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] Suggestions for reliable DID providerforCanada, USA and Europe
I've had some serious issues with Teliax as of late with their new Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility issues. I have had zero problems with their old servers. Interesting... I've got several lines on Teliax that have been in place for several months and the service has been very good. Recently we connected a new system to Teliax and I've been fighting the same issues you mention. I've been told the problem is with my software since SIP seems to work fairly well but not IAX. I also found out that my system is one of the first 20 systems to connect to their new Denver server. Now I'm curious about how many others are having the same problem. John ___ -- 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] Voted most stable and easy to use phone?
A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. We just installed a dozen of the Polycom IP-330 phones. Initially out of the box I wasn't real sure about the decision to use them. The phones are very small and don't seem to have very many features. However in use they have been great. They don't waste a lot of desk space, they don't overwhelm the users and they seem to provide adequate information. They're easy to use and Polycom reliable. The speaker phone is still really good though I'm not sure it is as good as the 501/601 phones. I haven't really done a side by side comparison of that but I think the 501/601 has a better speaker phone. I can't see buying another GXP after using these. The difference in price just isn't worth the aggravation. John ___ -- 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] ata device but for a soundcard
Adam Moffett wrote: So you want a device that will answer a SIP call, and play the audio out to a speaker? You're looking to build a PA system then? We achieved this using a Grandstream Budgetone configured to auto-answer, and just soldered a pair of wires across its speaker Neither the BT-101 nor the BT-102 support auto answer. I bought one to do this conversion only to find out after I removed the case that auto answer isn't available. The BT-200 is the one you want. John ___ -- 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 TDM400Pproduction: alternatives??
Steve Totaro wrote: 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. Steve, The way I read it, James is suggesting that the original poster would be better served to use the Digium card. While James is obviously related to Rhino cards, since he is suggesting that it would be better to use the Digium card, I find no offense in his post. Had he suggested Rhino cards that would have been a different story. I also agree that there is nothing cheap about Sangoma cards. However even as you mention, the original poster specifically asked about Openvox, so your suggestion of Sangoma cards is as out of place as your claiming James to be. At least James kept his suggestions to the cards of question. Is your issue with his remarks somehow related to your consultation business? John Faubion ___ -- 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] How to prevent logging of some entries in CDR
No one else is seeing this issue ? I have the same issue but I haven't put much effort into solving it yet. Too many other issues seem to get in the way. John ___ -- 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] WARNING[31046]: chan_sip.c:4978 process_sdp:Unable to lookup host in c= line, 'IN IP4 100101'
Well can you offer some explanation why T.38 faxing worked for months and then one day stopped working? Generally it is because some one or some process did one or more of the following: 1. Updated firmware on the ATA 2. Updated software on the server 3. Changed a configuration setting 4. Let the smoke out of a chip AFAIK none of these devices or systems are using artificial intelligence, yet. Therefore when something just quits you need to start looking for what changed. John ___ -- 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 1.4.16 and 1.2.26 released
The releases are available for immediate download from http://downloads.digium.com/. Could someone make sure the files are actually available BEFORE sending these out? John ___ --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 1.4.16 and 1.2.26 released
Could someone make sure the files are actually available BEFORE sending these out? I apologize for the way that sounds. It certainly sounded a lot more tongue-in-cheek in my head. John ___ --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 on Pcengines Alix board
I guess /tmp can live in RAM, but what about eg. recording ten-twenty WAV files to /var a day, and logs into /var/log? Do I have to worry about the card wearing out in six months? This is nothing really. Just make sure your using an industrial compact flash card. These support 1-2 million cycles where many of the retail cards only support 100,000 cycles. We also greatly limit the logs being generated. Writing logs files creates many times more write cycles than voicemail ever could. If your concerned about logs use syslog to send them to an external system. I'm not sure I understand the need for the PCI card to be perpendicular to the board. So I can use a flatter box. Then you don't want it perpendicular you want it parallel. The systems I'm using and even the e140 do this. There is a short riser card that plugs into the PCI slot on the board. The PCI card then plugs into this riser which allows the PCI card to be parallel to the motherboard. The cases I use are 12.5 by 10.5 by 2 so you can see that is a pretty thin box. John ___ --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 on Pcengines Alix board
Is the PCI slot large enough for full height, half length PCI boards ? Yes. Has you heard of a PCI Express version ? No but the way chipsets are coming down in price, I would imagine someone will have it soon. John ___ --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 on Pcengines Alix board
Thanks for the tip. It seems like they no longer manufacture them: http://www.neoware.com/products/hardware/ No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB RAM, and still has room inside the case for a hard drive. It is available without Win XPe starting at $339 new. The prices on these are coming down. - Fan-less, compact motherboard While some of these thin clients have fans, many of them only run the fan when set horizontally and thus don't have the advantage of heat induced air flow. - hard-disk (so I don't have to tweek Linux making too many writes and wear down the CF card) The newer CF cards are making this nearly a mute point. Seems like I provide updated software often enough that I never have CF cards wear out. We format a new image for the customer and send out a new card. They take a couple of minutes to power down the system, swap out the card and boot up on the new load. When done they return the old card to us for recycling and get a $10 credit. - a PCI card installed at an 90° angle (I prefer to use a PCI card instead of an external FXO gateway) I'm not sure I understand the need for the PCI card to be perpendicular to the board. I prefer the flatter box since they mount to a wall well and provide a nice compact installation. John ___ --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 on Pcengines Alix board
I'd rather use a PCI card to connect * to the POTS, and a hard-disk instead of a CF card. Do you know of a similar, small form-factor motherboard + case that would fit the bill? Many of the thin clients fit the bill nicely. I've been using MaxSpeed MaxTerm clients lately. Mainly because I bought several off ebay for about $50 each. These have a VIA motherboard that has a PCI slot, plus USB and IDE interfaces. They also have a CF adapter already in them and have room in the case for 2.5 hard drives. An added bonus is they are powered by a 12VDC adapter which makes battery backup a very simple task. John ___ --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] Grandstream GXP2020 + Asterisk 1.4.11
I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. I use on every phone the 1 as local port and in the rtp.conf From my knowledge of IP I don't think this is a problem since the address/port would be unique. However the example config I originally had from Grandstream indicated that each phone should use a different port and recommended to use the random port option on the phones. I have since assigned the port number on each phone to 1 plus the extension number. This was done to create a unique port number and to help with troubleshooting when using Wireshark or tcpdump. I set this in the config file for each phone. John ___ --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] RTP Read too short
Am I the *ONLY* one that has this issue? John Faubion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Faubion Sent: Thursday, November 01, 2007 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Read too short Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to correct this? I'm using SIPConnect from CBeyond and this appears on incoming calls. I haven't had any complaints about voice quality and I haven't seen any dropped calls. Should I be concerned? Thanks, John Faubion ___ --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] RTP Read too short
Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to correct this? I'm using SIPConnect from CBeyond and this appears on incoming calls. I haven't had any complaints about voice quality and I haven't seen any dropped calls. Should I be concerned? Thanks, John Faubion ___ --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] Advanced Dial Plan
Let's suppose that my sip extension 3000 want to call to (302).123.3211 I need a rule in extensions.conf to match with this number, right ? Let me see if I have this correct. You want to use the provider-302333-3000 for any call going out from 3000 and provider-30-3001 for any call going out from 3001. Basically a one to one mapping of extension to trunk, right? Try something like this... === exten= _X./3000.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) exten= _X./3000.,2,Hangup exten= _X./3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) exten= _X./3001,2,Hangup === This way if extension 3000 makes a call it uses the first one and if 3001 makes a call it uses the second one. John ___ --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] phone as control interface (was99bottlesof beer)
has anyone actually been satisfied with the performance of these powerline signalling devices ? Universal Powerline Bus is a vast improvement over the original X10. I believe the X10 devices used a 4V signal during the zero crossing point of the AC voltage to transmit 1 bit. Needless to say this made X10 slow, susceptible to line noise and not very reliable. IIRC X10 is only 75% reliable. By contrast, UPB is 20-40 times faster, it uses a higher signaling voltage so line noise isn't a big factor any more and has the advantage of controlling 250 times more devices than X10. This will help to prevent stray signals from the neighbors controller from accidentally controlling your devices. UPB is supposed to be 99.9% reliable with a latency of less than 100 milliseconds. Granted that still leaves a tenth of a percent of uncertainly. However that is without resorting to filters, couplers and the like. Granted in an existing situation there may not be a way to run more wires, but I evaluated them a while back and decided to stay away. You may want to take another look at them. Just like Asterisk has made great strides since the release of 0.7, UPB has brought the quality level way up. Of course this higher quality also has a higher price. John ___ --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] phone as control interface (was 99bottlesof beer)
Does anyone know of such a device that I can use over a network? It would be a pain to run a USB cable. I am thinking of devices that are like: I think your missing the key feature of these devices, UPB/X10. UPB and X10 are communication protocols that runs across the electrical wiring in the home. The 1132 box can be programmed using a USB cable to your computer but it doesn't have to remain connected via USB to control the lights and outlets using X10. Additionally you only have to connect one PC-UPB/X10 controller to control the other UPB/X10 devices. You will have to install more devices to make it all work but the commands come across the house wiring. John ___ --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] Grandstream GXP2020 / 2000
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 I have an installation right now in a real estate/mortgage company office with 36 GXP2000 phones. Average call volume is currently only about 150-200 calls per day but the number is climbing rapidly as they add more agents/loan officers. The latest firmware (Beta: 1.1.4.22) is a huge improvement over 1.1.1.14 though 1.1.1.14 is the current release firmware. Sadly some of the firmware loads we've tested have been horrible! The speaker phone is greatly improved. Call forwarding was an issue for several loads if using Asterisk 1.4 or later. Seems the SIP 302 message coming from the phone was corrupted. I had to hack chan_sip.c as a work around because this was a feature the had worked using 1.2 and was promised with the new system. Version 1.1.4.18 finally fixed that issue so the hack isn't necessary any more. The biggest complaint I have is the method of creating a config files for the phones. Unlike a Polycom which allows you to configure the phone using an XML file, the GXP requires you to create a text file with the configuration settings and then compile that file with their software. Additionally, if you perform a factory reset on the phone, it tries to connect to fm.grandstream.com/gs to update it's firmware load. So we are forced to run a caching name server with that address pointing to our own local server. (Don't even bother trying to tell me that all I have to do is change it in the web interface. After a factory reset, or a nice ESD zap which seems to nearly always result in a factory reset, the default address is back.) We have approached the configuration issue several different ways. The current method is using a MySQL database. We built the database and then modified the HTML from the phones web configuration to use it to update the database. We use a cron to monitor the last update time and generate a new set of config files once the database has been updated. If you only have a small installation or very little turnover, our previous method of using a text file for the database and a perl script to update the files is probably sufficient. While I haven't gotten any complaints about the cheap toy like feel, I think this is mostly due to lack of experience on the part of my users. With the GXP being the only VoIP phone they have used, they do not have a basis for comparison. The original quote offered Polycom, Aastra, Snom, and Linksys phones. The GXP was chosen strictly by price since the price difference saved them over $1000. I now demonstrate the phones on a portable system to allow the customer to see and feel the difference in the phones. I have also increased the price of the GXP phones I sell. Between these two measures I don't sell as many GXP phones. I feel the increase in price was justified based on the additional work I've had in using the GXP phones. Since the bugs are mostly worked out now, these should be more profitable for me in the future. John ___ 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] sip.conf best practices?
The obvious alternative is to use the extension as the sip UID: Use the extension as the UID and add the mac address as a comment. Like so: [123] ; Joe Smith ;mac=000E08DA0409 secret = blahblah ... and so on and so forth This will give the best of both worlds. The mac is readily available and the dialplan is clear. I usually try to go one further and setup dhcp to set the last octet of the IP address to the extension number. This makes it easy to point a browser to the phone for configuration as well. John ___ 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 licenses installed - voicemail has no audio...
I got my G729 licenses installed.I can make calls out and receive Make sure you add g729 to the voicemail config as well. John ___ --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 licenses installed - voicemail has noaudio...
I can't even hear the password prompts. Ahh... have you loaded the G729 sounds? Are you getting errors in the logs? John ___ --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] phone directory with asterisk
To prevent further missunderstanding please do not refer the SI-120 as a snom phone. If you need support please contact snom India. Tim, If it is sold by snom India, and one is to contact snom India, I can certainly see how one could infer that it is indeed a snom phone. John ___ --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] phone directory with asterisk
I can only pointing out this issue, trying to prevent bad user experience with these phones. I'm very very unhappy about the situation but it's out ouf my reach to change this. Yes, and I certainly understand how difficult it can be to get upper management to understand that their cost cutting/joint venture baby is also cutting into customer satisfaction. One can only hope that you can get through management before it takes too much toll on the company. Good luck! John ___ --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] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) John ___ --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] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) And received at 10:58 CDT (16:58 UTC)... May have more to do with where it is sent from than the list itself. John ___ --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] Learn some terminalogy before mounting thistask.
They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes, and just to complicate matters further, they are probably asking about the NT-1 or NT-2 which is the Network Termination type. NI1/NI2 usually refers to the National ISDN phase, for which the difference has generally been eliminated. The NT-1 is a 2-wire U interface while the NT-2 is a 4-wire S/T interface. Since both NT-1 and NT-2 use the same RJ-45 connector, albeit different pins, most cards now support either interface, and often auto-magically. I haven't used the Sangoma cards, but since the Digium cards support this, I would expect the Sangoma cards to do the same. Now the reason they are asking is, if your router only supported a 4-wire S/T interface, then they will need to provide a CSU/DSU to convert the NT-1 or U interface. Or if you already had an external CSU/DSU you might want the NT-1 interface. Now having said all of that, I'm actually going to recommend that you tell them you need the NT-2 interface. Why? Demarcation. Most providers want to provide the CSU/DSU and give you a S/T interface. The reasons are, if they provide the CPE interface, they are going to use a box they know will work. That way if you have problems, they can confirm that the circuit is good to that point. What you do with the circuit after that isn't their concern. Plus if you were to accidentally connect the 4-wire circuit directly to 120 volt AC power, you are most likely only going to fry the CSU/DSU and not the equipment in their concentrator around the corner. Additionally, it works both ways. If the cabinet on the corner of the block takes a lightning strike, the CSS/DSU goes up in smoke and hopefully protects your equipment. So just because you can use either interface, doesn't mean you should. Clear as mud, isn't it? John ___ --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] Transfer Call to Cell Phone
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call toll
bridging which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that ATT is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, You must first dial a 0 or 1 before calling this number or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.
The only way I can get it to work is by have the call on the 1st
line then transfer it out on the 2nd line. After that is complete both
lines are free.
Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example.
In extensions.conf use something like this.
[global]
SIP-PROV = sip.urprovider.com
; Now set the call forward numbers
CFN21 = 551234 ; These are normally set in an external file
[internal]
exten = 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})
[macro-stdext];
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - Our call forward number
exten = s,1,Dial(${ARG1},10)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}0]?s-CFWD,1)
exten = s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten = s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten = s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = s-CFWD,2,Goto(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,2)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not answered in 20 seconds, then we send it to voice mail.
Finally if the user presses the star button during the attempt, we send them
on to Voicemail mail so they can check their messages.
Hopefully this helps.
John
___
--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] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.
When making an outbound call, the outbound peer return a 301 forwarded with URI to other domain, but asterisk think it's a local domain and try to look it up from extension.conf. What phones are you using? This sounds a lot like a problem, I have using Grandstream phones. John ___ --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] kore dump
What is a god Windows application to read core dump files? Microsoft jokes aside, I would seriously doubt there could be a good Windows application for analyzing core dumps. Due to the OS specific nature of core dumps, the need to have the source files, debugger and more, would make it difficult. I'm not saying there isn't one, I've just never heard of one. When developing software modules, I've had some success using crash on Fedora systems. Though as a whole system, a review of the logs to see what was changed just prior to getting the core dumps has been more effective at isolating the problem than the analysis of the core dump. John ___ --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] Transfer Call to Cell Phone
by the way selling does not depend on the amount of lines you have and we are very productive trust me True, very true. There are lots of very productive sales people that don't need a phone at all. From the paper boy to car dealers, lots of sales don't require many phone lines. Of course at the same time, a typical call center wouldn't be very productive with only two lines. I have seen a million dollar corp work off four lines so your statement is quite vague... We have a few agents that have million dollar months and even a couple that have had million dollar weeks! But that isn't the point, is it Otis? The problem is you got your feelings hurt because instead of reading my reply, you assumed that I was putting your company down. My first paragraph was kind of a open thought process so that you and others might comprehend the basis of my reply. What I was trying to wrap my head around, was just how productive a system with only two lines could be if a single call came in and was then routed back out the other line to the outside sales guy. Now if you were using a digital line, perhaps we could consider using the signaling to redirect the call from the originating source directly to the salesman's phone and thus free up the lines for the next call. But no, you said pots lines as in Plain Old Telephone Service (POTS) which means we don't have the option of using some fancy, out of band signaling to redirect the call. So my thinking was, as I said before, Surely you have more than two lines. In my twenty-two years of telephony experience, dealing with everything from single line phones to key systems to PBX systems to Nortel DMS-500 switches, I only remember one sales office that only had two lines and that office was literally an 8 foot by 8 foot closet with two phones and all calls were outgoing. Yes, my answer was a little vague. So was the information you provided. Now had you bothered to read the 2nd and 3rd paragraph, you might have noticed that I provided a few methods that you could consider. My intention for doing this was simple. Maybe one of the ways mentioned would spark a response from you that would help to clarify the right way. Now suppose for a moment that you had actually read the reply. Let's also pretend that in reading it you realized that, yes, you have two pots lines, but what you had meant to say was that you had two unused pots lines along with some other form of incoming trunks. Then maybe you would have responded with an email to clarify that, to which I could have suggested that maybe you could look into a two port cell phone gateway to keep the incoming lines free and still keep connected to your sales guy. Can you see how we could have used that information to consider the right option? Considering that this list is for non commercial discussion, our only form of payment here is in the repayment of our debt to others that have gone before us and helped us out. Next time please appreciate the fact that someone else took time out of their busy day to consider and to reply your request for information. Now if you would like to provide a little more detail with your request, I'm fairly sure that someone here will likely respond to it. John ___ --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] Inexpensive Layer 3 Switch?
Any recommendations on an economical layer 3 switch? I've been quite happy with the Netgear FS728TP ProSafe switches. These are 24 port 10/100 switches with 4 GigE ports and PoE. They also support 1000-SX GBIC through an optional module. The total PoE budget for all 24 port is 195 watts. We run 38 GXP2000 phones on a pair of them. We have one with 19 phones and a DLink DWL-3200AP wireless access point drawing power and the current load is under 75 watts average and has peaked at 97 watts. The other one has 19 phones, two Ethernet cameras and draws even less. All for less than $375 shipped. They even have a $25 rebate on them until the end of the month. Plus they even have a Lifetime Warranty. One of the cool features I discovered after installation was the built in Time Domain Reflectometer. The TDR is great for testing out the cables right after installation. We were able to use it to locate the screw that the drywallers intentionally ran through one of our cables. It was so obvious that the contractor paid to replace the cable. The only negative comment I have about them is the drone of the fans. Our wall mount rack is in the break room and the switches are easily the loudest item in the rack. They are not as loud as the Dell server we originally bought so if your using a 1U server from Compaq, Dell, HP, etc... you'll be fine. John ___ --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] Transfer Call to Cell Phone
Polycom Phones 1. New call 2. Press 9 access outside line 3. Dial Cell Number 4. Transfer the call that way. Once you initiate a new call you will tie up the second line. Your asterisk box will now be bridging the two lines. The lines will stay tied up until the salesman drops the call. One method you might be able to employ here would be to add a call transfer to the pots lines. Then you would need to send a hook flash to the pots line, and dial the salesman's number when you get the dial tone. Then, depending on how your local Telco supports the call transfer feature, you may be able to free up the line. Not all Telcos support this the same way as some consider it a method of toll avoidance and thus drop the call. This would be possible in an area where a call from party A to party B is a local call and the call from party B to party C is a local call but a call from party A to party C would be a toll call. Since the call from party A to party C is a toll call, the Telco may opt to drop the call. If the transfer part works, there may even be a way to setup the dial plan to intercept your phones call transfer feature and use a 1-2 digit code to select which phone number to send out. I have not done this but I think it is reasonable as I've heard of home users doing it. By the second option, are you talking about the TDMA/GSM gateway? If so, yes this is pretty slick. We considered it initially as well. Our decision not to use it was based on the fact that many of our agents are on different mobile plans. I think when we requested the info from the agents we had 6 different wireless companies represented. Since Sprint/Nextel, Cingular/ATT, T-Mobile don't like to play nicely with other, we didn't expect to see any real savings from the free mobile to mobile calls. This was mainly due to the fact that we don't pay for the agents phones and thus we can't really tell the agents which carriers to use. I do know of a couple of installations where the company does provide the phones and I understand the savings can be significant. I was told by friend that the box they installed paid for itself in just a couple of months. But their phone were already on the same plan. John ___ --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] Inexpensive Layer 3 Switch?
If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. What model did you use? I've been very happy with the FS728TP as I mentioned earlier. I haven't had any problems so far. Granted I haven't had to call Netgear so I don't have anything on which to judge their service. I will say that unless your dealing with a very small system, you should probably steer away from the FS726TP. It only supports PoE on the first 12 ports and doesn't have anywhere near the features of the FS728TP. John ___ --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] kore dump
I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. Sure! You should already have the required script. Just run it from safe_asterisk. Here is a link with more info: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs -html/x389.html John ___ --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-users Digest, Vol 35, Issue 89
Dude you got to be freaking kidding me - are you really sending this email to everyone who posts on the Asterisk list? No, most likely he has an autoreply/vacation/out-of-office message enabled. I would expect us to get more of them. Just be thankful he is on digest mode! John Faubion ___ --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] Transfer Call to Cell Phone
I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? Do it right? You really haven't provided enough information to make the right decision. Do you have more than two lines? Surely you have more than two lines. You mention his extension and the main sales extensions. I can't imagine a sales department with only two lines. Well I can, but they don't sell much! 8) If you have other lines available, such as through an ITSP, T1/E1, or etc, then you only need to map his extension to an outside line. This could be done either through a follow-me, call forwarding, fixed routing, or etc. As an example, we have several agents (we're a real estate brokerage office), that only come into the office occasionally. Since most of them use their cell phones for nearly all of their business, I have fixed routing to send calls to them. I will soon have an IVR for them to be able to change that fixed routing on their own. We also have some agents that have a regular desk here in the office. For them, the use call forward unanswered at the phone to route the calls to their cell phones when they are out of the office. The owner uses follow-me to route her calls to the office phone, her home phone and her cell phone. Another way to do it would be to install a SIP/IAX/TDM to TDMA/GSM gateway. Make sure the provider is the same as the salesman's cell phone provider and your mobile to mobile minutes can be free. If you have more than a couple salesmen, this route will likely entail a multi-port gateway but the idea is still the same. As far as the right way, that depends on way to many factors tat you haven't addressed. John Faubion ___ --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] Asternic Flash panel
Anyone get it working on 1.4. Checked out their website no updates for some time now... Sorry for the late post. I'm tying to catch up on my email. I have three systems using it. One with 1.4.3 and the other two on 1.4.5 if you still need help. John ___ --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] chan_features.so / asterisk 1.4.5
Carsten Bock wrote: See the 1.4.5 Changelog: (http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.5) 2007-06-07 19:46 + [r68196] Olle Johansson [EMAIL PROTECTED] * channels/chan_features.c: Disable chan_features by default in menuselect Well isn't THAT enlightening?!? Anyone have any idea why it was changed? John ___ --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
