Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
On April 1, 2009 01:32:18 pm Jason Aarons (US) wrote: I don't think a off the shelf modem has the necessary DSPs to convert voice to codecthat is why a Voice Gateway/Analog Telephony Adapter or FXO/FXS cards exist instead of modem having a second life. There are no DSPs in any of the telephony products you get from Digium, Sangoma, etc., with the exception of the VPM (echo cancellation) add-on modules and some minor hardware assistance that hardware that the digital span cards provide. In fact, that was the entire point made by Zaptel and also by projects such as SpanDSP -- you don't need a fancy DSP to do mid-scale voice processing on the computers of the day, and processing power is only increasing. -A. ___ -- 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] iphone, skype and asterisk ...
On March 30, 2009 12:48:59 pm randulo wrote: Except for roaming and in particular international roaming, isn't the best plan to forward calls the iPhone. It is a phone, too isn't it? Or just a game platform, browser and GPS? That's pretty much what I do; I use siax (I have a jailbroken iPhone) and it works great. I use it for testing, and when I am in friendly wifi areas and want to talk to someone without using my minutes. As cool a device as these things are, the all in one approach has rarely worked well. Seems to me a good phone should be a good phone. Maybe part of the future is netbooks with 3g, too? Actually that would be a great idea. I already tether the phone to the laptop, and have an HSDPA card for the computer for when tethering just isn't sufficient. If I could get a bluetooth keyboard hooked up to the phone and the phone had slightly higher resolution, it'd be an almost perfect piece of convergence kit. -A. ___ -- 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] 2008 Post Count
On January 2, 2009 01:44:14 pm David wrote: 2007 2006 Andrew Kohlsmith 290 2005 Andrew Kohlsmith 731 Damn... I'm slipping! 2nd place in 2005. -A. ___ -- 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] top posting again [was: Re: CDR Design]
On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote: To me top posting is like people talking about SIP Trunks. There is no such thing as a SIP Trunk. There are SIP connections, peers, friends, etc. The term is simply a marketing buzzword to make people that don't know much about VoIP feel all warm and fuzzy about a product. I thought the term SIP trunk came from old PBX-heads trying to apply the term SIP to a destination route, much like LD trunks, POTS trunks and even remote office trunks. -A. ___ -- 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] top posting again [was: Re: CDR Design]
On December 17, 2008 06:59:19 pm David fire wrote: you are soamming my mail box whit this useless discution the solution is doble posting (top and bottom) It's a public mailing list. If you're having trouble managing it, you may want to try a digest version, or perhaps a moderated list. -A. ___ -- 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] Rate My Dialplan Contest Announced - Win a Phone or Copies of APSTel Visual Dialplan Std or Pro!
On December 4, 2008 08:31:40 pm Matt Gibson wrote: 1st place: An APSTel dial plan (professional license) donated by -- you guessed it - APSTel! 2nd place: An Aastra 57I IP telephone donated by Ottawa Phone Systems and Flewid Inc! 3rd place: An APSTel dial plan (standard license) donated by APSTel! So... if you can write the slickest dialplan, you get dialplan generator software? -A. ___ -- 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] cepstral vs festival
On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- 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] cepstral vs festival
On December 4, 2008 02:14:52 pm Erik (Caneris) wrote: Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Definitely would be cool, you don't lose any ad revenue and I don't have to use up my minutes. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. You have Kitchener/Waterloo! Yay dials Oh. No traffic. Boo-urns. I'd definitely like to know when you start populating the traffic part of K/W (and separate out london, it's a poor choice to group. Kitchener/Wwaterloo/Cambridge sure... but London? That's a common Torontonian thing to do. :-) -A. ___ -- 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: What do you guys think of this?
On December 1, 2008 07:21:33 pm Doug wrote: Hmmm. When our users are pounding the network with BitTorrent traffic, we just shut them down and wait for them to complain. It's against our Acceptable Use Policy, and causes all sorts of VOIP headaches. As someone who is the technical lead for several ISPs, it is my professional opinion that you haven't a clue how to run such a thing. Torrent does not interfere with VOIP on a well-designed network any more than FTP or web browsing. Honestly, hire a competent admin to set up and run your infrastructure. If torrent's killing VOIP, that means that adding more VOIP will also kill it. Or excessive web browsing. Thank God I'm not one of your customers. -A. ___ -- 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] network design philosophy and practice
On October 29, 2008 10:19:36 am Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? I'm a pragmatist; most offices have one network jack at each station; I run voice and data on the same physical wire, but if at all possible I try to split things off using smarter switches and VLANs. -A. ___ -- 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] fax / t38 gateway
On October 28, 2008 12:58:25 pm JD wrote: The folks that devloped the fax V.protocols took into acount typical copper problems like noise or echo. But what they never conceived of as even being possible is that a call might shift around in the time domain. Thanks to jitter/latency, the delay time of a call can change in the middle of the call. That isn't possible with copper technologies. This makes faxing over even G.711 a dice roll. A good description, but please, refrain from copper technologies -- copper has nothing to do with it. You're talking about conventional PSTN (circuit switched) technologies. -A. ___ -- 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] Fring: Open VPN client to be installed on the mobile, which mobile?
On October 27, 2008 02:01:43 pm Jeff LaCoursiere wrote: Speaking of fring, I just got my brand new iphone 3G. Anyone have any comments on how well fring or any other sip client (siphon?) works on iphone? I do not like fring. It's buggy, it's unstable, it looks goofy -- but I have to say that yes, the SIP client appears to work. It won't reinvite off of their servers, though, so your audio path goes through them all the time. I need to learn how to write iphone apps and just write a simple straightforward SIP phone for it. -A. ___ -- 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 clients for smart phones?
On October 3, 2008 04:15:26 pm Tariq .. wrote: it is FRING i'm sorry for the mistype... www.fring.com I just downloaded it for the iphone... it's pretty cheap looking, crashes occasionally and appears to force all audio through their server, but I have to say that yes, it does have potential. Thanks for the pointer. -A. ___ -- 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] OT: Re: sip clients for smart phones?
On October 3, 2008 08:56:34 pm Philipp Kempgen wrote: I could live with 1 or maybe 2 of these issues but 5 is a bit much. You didn't even notice these problems, so, ok, sorry for being rude. But for people who are used to email in ages it feels like a punch in the face. It's a real culture clash. Having been a user of email and a staunch advocate of text-only messages, minimal signature lines, proper trimming and bottom posting for well over 15 years, I have to say that I've never felt punched in the face nor experienced any kind of culture clash. I think that perhaps you're too sensitive to these things. If you really have been using email for as long as you claim, you may want to implement killfiles so protect your thin sensibilities, or perhaps grow a thicker skin. And while I don't like webmail or Hotmail either, my MUA seems to read his email just fine; perhaps you should upgrade to a better client yourself? :-) -A. ___ -- 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] OT: text/plain (was: Re: Re: sip clients for smart phones?)
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote: Thunderbird could probably render his text/html part just fine but I don't want it to. (Nothing is wrong with preferring text/plain in the MUA.) Thus it renders his text/plain part which lacks line breaks. I posted some links to the list archive to demonstrate that the problem has nothing to do with my MUA. ---cut--- http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html ---cut--- That quoted text is not very eye-friendly. Konqueror also renders these as huge one-line messages; I am blaming the mailing list archival software for this, as it is not being sufficiently suspicious of the data it's processing; it should be either stripping the pre tags or otherwise forcing them to be web-friendly, IMO. -A. ___ -- 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] OT: Do You Know What the Problem With CDMA is?
On September 25, 2008 09:01:52 am Dean Collins wrote: Yep you got it world coverage includes all the countries of the world like USA, Canada and Mexico, and not something like USA and 212 other countries globally. BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it into a funding bill for the war that they had to use CDMA to 'support usa businesses'), of course that means that they now use a different handset type to all of their neighbours... though I hear Iran will also be forced to implement CDMA once they are 'liberated' which should be any day now :) That doesn't mean that GSM towers won't be built, it just means that the CDMA towers will be there first. I dunno; GSM 3G is all CDMA tech anyway. -A. ___ -- 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] OT: Do You Know What the Problem With CDMA is?
On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A. ___ -- 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
[asterisk-users] PRI incoming call forward / call redirect
Good morning,
I have a Bell Canada PRI here (switchtype=national) and I am trying to perform
a call-forward-unconditional on one of the DIDs.
The idea is that when DID 5551234 receives a call, Asterisk redirects it back
out the same PRI to some external number.
This is simple enough to do with something along these lines:
[PRI]
exten = 5551234,1,Set(CALLERID(RDNIS)=${EXTEN})
exten = 5551234,n,Dial(Zap/g1/5556789)
This is a brute-force approach but there are two problems:
1) it's not a true call forward
2) RDNIS does not appear to be getting set (i.e. the remote box with 5556789
as a DID does not seem to see RDNIS
I'm not overly concerned about 2BCT capability at this time (it *is* talking
to a 5ESS although I'm not sure if Asterisk will attempt 2BCT with national-2
switchtype) but it is important to be able to retrieve RDNIS, as the hope is
to redirect a number of DIDs to one external number, and have the external
number see which the original number was through RDNIS.
I had this working great the other way -- some external POTS number
call-forwarded with *72 to a DID on this PRI, the DID saw RDNIS just fine,
but now I'm trying to go the other way round.
Any ideas? This is Asterisk 1.4.18.
Regards,
Andrew
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Re: [asterisk-users] Semi-OT Satellite?
On August 23, 2008 07:57:33 pm Alex Balashov wrote: Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at all over high RTT latency (= 400 ms). Why not? Is there some kind of timing involved in encapsulating data that I'm not aware of? -A. ___ -- 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] Intermittent T.38 pass through
On August 11, 2008 06:59:23 pm JR Richardson wrote: So my question is this: Can I setup Asterisk to only allow t.38 pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Can you use disallow/allow with UDPTL? I'm not sure, I've never played with this before. -A. ___ -- 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 behi nda MeridianNorstar
On July 24, 2008 04:42:42 pm David Cook wrote: Have the Norstar programmer send all 3 digit, unused extensions to the PRI. Then Asterisk will see 221, etc. and can handle at your dialplan sees fit. Yes, this works, but you won't be able to treat those as regular extensions; the Nortel will treat them as external numbers. -A. ___ -- 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] MagicJack quality
On July 17, 2008 11:44:07 am Dean Collins wrote: 1/ RD costs v's number of units manafactured per annum. That's bullshit; There are many more office phones than office desktops out there, and the research has been paid for many times over. Think of how long the Meridian 1 has been around. 2/ Retail pricing the markets will still purchase at. That, my friend, is the right answer. :-) -A. ___ -- 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] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
On June 23, 2008 08:08:53 am OCG Technical Support wrote: I little more digging and I confirmed that cell phone VM and FAX waiting icons are in fact controlled by a proprietary SMS message format. Here's what I found: Yes; this is the same sticking point I hit; you can't use an SMS email gateway, because the SMSC mangles the message format in order to prevent this and other hacks. (Imagine anyone sending you an email and controlling this and other things on your phone!) Your best bet is to try and find an internet SMS company that can control the MWI for your provider. -A. ___ -- 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] OT: Re: OT How Digium Saved My Bacon!
On June 17, 2008 01:45:43 am randulo wrote: The screwdriver is reversible, it swings both ways, pull out the shank and stick it in the other way, it becomes a Phillips. I'm tellin ya, there Digium engineers are good! Most every pocket screwdriver that is sold as a promotional item is like that. It's not always good; I cut my hand pretty badly when the phillips end slid clean through the screwdriver and into my hand once. I wonder if they'll consider a slot/robertson combination for us northerners. :-) -A. ___ -- 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 How Digium Saved My Bacon!
On June 15, 2008 12:04:01 pm randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! Before you ask, it's 1.0 I think. ? -A. ___ -- 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 How Digium Saved My Bacon!
On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- 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] 911 via MAX TNT ??
On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... -A. ___ -- 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] Help Please - Asterisk MYSQL interface seems to be eating data
On May 5, 2008 01:58:42 pm Tilghman Lesher wrote: Hmm. Haven't found any Digium Stockholm office to discuss with ;-) That hasn't stopped any of the Canadian employees. :-) That's because nothing stops Canadians, short of Hockey Night in Canada :-) -A. ___ -- 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 Card: Power Connector, from SATA to NORMAL
On May 4, 2008 07:24:45 pm Rob Hillis wrote: Customer's insistence. We didn't have a choice, really. Nothing wrong with that, it just adds more billable hours. :-) -A. ___ -- 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 Card: Power Connector, from SATA to NORMAL
On May 4, 2008 08:40:10 pm Jay R. Ashworth wrote: Customer's insistence. We didn't have a choice, really. Nothing wrong with that, it just adds more billable hours. :-) As long as it does. I don't know about you, but whenever a customer wants me to do work and does not want to follow my recommendations, I have the paper trail copied out in triplicate just to cover my ass. Sometimes they're right, but generally when they ask me to do something they are asking me because they are unable to do it themselves, so I am extra-cautious when they won't follow my advice. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New generic sounds
On May 2, 2008 03:13:40 pm Norman Franke wrote: enter the four-digit extension of the person you are trying to reach I would suggest breaking that up Please enter the digit extension of the person you are trying to reach then you can use the individual numbers and fill in 2 digit, 3 digit, etc. -A. ___ -- 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 PRI card hi-Z for sniffing?
On May 1, 2008 11:39:52 am Tony Mountifield wrote: Does anyone know if the Digium PRI cards can be configured or modified to have a high-impedance input on the RX pair? I would be interested in this in order to build a bi-directional PRI audio sniffer using two E1/T1 ports per trunk to be monitored. I looked in to this exact thing several years ago, and while I do believe that the hardware is capable of it (I don't have a card in front of me, but I believe the T1 termination is done inside the QuadFALC), the driver does not have this capability. I don't think it would be impossible to add, but it would take some work, and please keep in mind that this was a number of years ago that I did look at this on the TE405P, so my memory may be a little hazy. -A. ___ -- 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] cdr_custom outout to serial port
On April 12, 2008 03:12:31 am Col Ferguson wrote: Hello, I have a system in a motel that needs call billing data output through its serial port so the existing motel management software can collect the call billing info. Is there any easy way to redirect the data that goes into the cdr_custom/Master.csv file to go out the serial port ? I've written a few variants of what I call a CDR MUX -- a little application that reads CDRs from a legacy system via serial port, reads CDRs from Asterisk, combines them and outputs them in a specific format out another serial port for some third-party billing system. Essentially that's what you're asking for, minus the legacy system part. I imagine the billing system isn't interested in a straight tail -f /dev/ttyS1 output, so some level of translation would be required. -A. ___ -- 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] Where is the Digium DS3 card?
On April 7, 2008 02:01:08 am Alex Balashov wrote: A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex too, perhaps. Haven't tried to see how much it can handle when TDM-RTP translation is required. I'm curious; are the cpu/tdm/dsp requirements for 672 g729 rtp streams that much higher than 672 v92 data streams? I have done work for a dialup ISP that has probably 20 of the damn things running for quite some time now with zero issues, and I can't imagine that the RTP requirements are higher than v92's. -A. ___ -- 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] Where is the Digium DS3 card?
On April 6, 2008 11:12:33 am Steve Totaro wrote: I cannot recommend the Adtran MX2800 M13, it has redundant everything and is very easy to setup and not very expensive either. Agreed; I've set these up and they are rock effing solid. We did have a shelf controller die and without the backup shelf controller, we had our only DS3 down for several days... the replacement shelf controller was lost by UPS... talk about a train wreck learning experience. :-) Ahh, the good old days... -A. ___ -- 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] Slightly OT: Getting VOIP number into phone book
On March 25, 2008 02:15:42 pm Lacy Moore wrote: I think that is one of the biggest things that businesses overlook when switching to Voip. It's hard to get in the directories. I have to say that it's been many years (well before voip) that I've gone to the directories. Google and yellow pages for pretty much everything. White pages if I know *exactly* what I'm looking for, but even then I tend to throw it in google before I go hunting for that big damned book. -A. ___ -- 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] Unable to obtain dialed number through ZAP
On March 24, 2008 02:38:03 am mark morreny wrote: What I need to do is to try to route called based on the dialed number as I have multiple DIDs on my line. Is this something that can be done? Is this something to do with the hardware that I am using? If so, what kind of hardware do I need to accomplish this task? As others have mentioned, if you have an analogue connection (and even in some cases, a digital one), this information is not available to you. A properly configured PRI or BRI should get you this data, and if you have a specifically-provisioned CAS T1, you can get it. -A. ___ -- 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?
On March 20, 2008 02:33:52 pm Anselm Martin Hoffmeister wrote: Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen: And what happens if at the time of the shutdown there was a ROTFL Trafrir, you made my day. Oh god, I didn't realize that wasn't a typo until you wrote that... Very well done, Tzafir. Professionally executed. -A. ___ -- 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?
On March 19, 2008 12:43:21 pm Bill Andersen wrote: I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. If you're continuously restarting Asterisk, there is something wrong with your setup: hardware, software or both. I have many installs out there on commodity hardware (either pure-voip or digital (PRI) only with Polycom handsets) and none of them need to be restarted. Now we're not using queues; straight extensions with voicemail, some paging and followme, a little CTI (click to dial), and a 24h page the poor shlub wearing the pager this week for emergency support. You know, pretty standard systems; the kind of thing I'd think any small business would have. None of these are PoE, have separate switches or special VLANs or anything like that. Think of what a small 5-50 person office would have the money for. I hear this complaint from time to time, but I've never really sat down and thought about what could be causing it. Which version(s) are you running? Whose hardware, what linux distro, are you running FreePBX or straight-from-sources Asterisk? I'll take you on your word that you're not trolling. Let's dig in a little. -A. ___ -- 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?
On March 19, 2008 07:00:20 pm Steve Totaro wrote: I would not consider a Dell SC440 w/RAID 1 Server Grade you can pick them up for $250 on sale. Why not? Is the price not high enough, or is there some technical reason? I ask because your only explanation as to why it's not server grade appears to be the price. I've got no idea what a SC440 is, can't be arsed to look it up, but your post seems to indicate that expensive must mean good quality. -A. ___ -- 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?
On March 19, 2008 05:05:05 pm Bill Andersen wrote: CentOS release 4.4 (Final) Kernel 2.6.9-34.0.2.ELsmp (SMP) Asterisk 1.4.16.2 Dell SC440 w/RAID 1 Digium TE120P The GUI is a commercially available product, to remain un-named at this point. Ok, and what specifically are the types of problems you are encountering? choppy audio, dropped calls, stuck calls, kernel panics, asterisk crashes...? -A ___ -- 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] IAX complaints? What are they?
On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote: With SIP you can attach custom variables to calls (using X-... headers). IAX (Inter-Asterisk eXchange!) can't do that (yet). With IAX2 you can share variables too. I believe Tilghman had supplied a patch to do exactly that several months ago. -A. ___ --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] 'Traditional' Faxing
On Monday 12 November 2007 07:54:42 Dave Fullerton wrote: From what I've heard, I think your best bet is to buy a multi-port T1/E1 card for asterisk, put your E1 in one port and a channel bank in the other port, then plug your fax extension into an FXS port on the channel bank. Since both legs of the call pass through the same E1 interface card asterisk can bridge the call on the card itself and the timing issue should become moot. I have not done this nor have any hands-on experience to share, but I have done some research into this in the past. This is also the method Fonality recommends for customers of their asterisk based system: That is exactly what I do in my systems. I did have success with a TDM400--T100P but that was a long time ago and I was unable to repeat it recently. I didn't spend a whole lot of time on it, though. -A. ___ --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] PSTN failover
On Tuesday 16 October 2007 03:49:37 Atis Lezdins wrote: Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only thing that continues is h extension. You must of course use 'g' in the Dial flags so that it continues on in the dialplan after hangup... -A. ___ --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] CDR
On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote: Michael Collins wrote: I don't know if it's relevant or not, but I do know that at least one legacy PBX vendor (NEC) has a 'solution' that helps with some of the sillier CDR's that could get generated. They have what they call a pseudo-answer timer which is basically just a way of saying, If a call doesn't last for at least X number of seconds then it really isn't a call and no CDR should be generated. It is a bit of a case of throwing away all really short phone calls, even legit ones, but it does also get rid of the silly stuff: I pick up, get dial tone, then hang up or I pick up, dial ext 1234, let it ring for two seconds and then hang up. I would definitely want a CDR record in this case. Out of curiosity... why? Both you and Atis seem to want to see CDRs for non-calls, and I'm unable to see why. -A. ___ --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] PSTN failover
On Monday 15 October 2007 19:50:03 Philipp Kempgen wrote:
I'd basically just Dial() 2 times:
Dial(SIP/...);
Dial(Zap/...);
No need for priority jumping. And not need to check if
the ChanIsAvail(). Just Dial().
Why not just do it the correct way?
Dial(SIP/,,g)
GotoIf($[${DIALSTATUS} = BUSY]?busy)
GotoIf($[${DIALSTATUS} = NOANSWER]?noanswer)
GotoIf($[${DIALSTATUS} = ANSWERED]?answered)
Dial(Zap/...)
Of course, I do this inside a macro, and I emit correct CDR and correct
hangupcauses for those who use my system.
Dialing twice like that without checking your return value is an invitation
for future problems.
-A.
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Re: [asterisk-users] really sorry about this - E1 vs T1
On Monday 15 October 2007 17:18:00 Andreas van dem Helge wrote: On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use the jumper settings. Seems like a bad design. Why not just make it a software choice?? Did he not just give you the software choice? -A. ___ --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] DS3 Interface
On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. ___ --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] DS3 Interface
On Tuesday 09 October 2007 14:32:38 Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm And your point, precisely, is what? Someone who has a criminal record can't be a technical authority? Someone can't have a criminal record without being a scumbag? Or perhaps that you prefer to write off those who can best your technical prowess by any means possible? My money's on the latter. -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? I don't know about you, but I've had nothing but very good results with VOIPSupply. I didnt do huge business with them, but I have purchased new and refurb polycoms from them without so much as an ounce of pain. -A. ___ 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] where is 1.4.12?
On Thursday 30 August 2007 9:49:57 am Matt wrote: I want to reply to this my initial comments were not trolls. I think, however, my initial comments reflect what alot of the asterisk community is experiencing.WE support asterisk for people. WE also sell phone systems based somewhat around the asterisk platform.WE run several asterisk softswitches to support our VoIP infrastructure. That being said, we have several folks who know asterisk almost like the back of.. something (does anyone really know what the back of their hand looks like?). It's rather embarrassing for us to have to say to a client.. well it looks like the version of asterisk you loaded has a bunch of bugs.. why don't you roll it back to this version? CLIENT: But but.. it's the newest, shouldn't it work? It just reflects poorly on Digium/Asterisk in general. If that is how you're selling and supporting Asterisk systems I think your business model is flawed. While I won't argue your other points, I think that if you've been working with Asterisk this long and you hit this kind of problem more than a few times then it's clear to me that you do not in fact know Asterisk as well as you thought! I'll admit I've been bitten once or twice by bugs AFTER a rollout, the vast majority of my installations work, as far as the customer is concerned. -A. ___ --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 it posible for an incoming to ring to Polycom and cell at the same time?
On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. I'm not sure where you are getting that assumption from, as I have been Dialing Zap/fooZap/bar, SIP/fooSIP/bar, IAX/fooIAX/bar and combinations of all three for the past several years. The only trick, as Anthony already showed, is to use 'r' when dialing a cell phone so that the caller hears the expected ringback, and not the carrier's The cell phone you are trying to reach is out of the area messages if the cell is out of range or off. -A. ___ --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] RAW asterisk!
On Thursday 16 August 2007 2:57:06 pm Barry L. Kline wrote: As far as tutorials, just pick up a copy of Asterisk: The Future of Telephony. Most of the howto for compilation is there, albeit somewhat dated until the newer version of the book hits the press. I'd wait a couple of weeks, the 2nd edition just went to print, so wait for it. I know two of the three authors personally, and know all the hard work that went into bringing the 2nd edition up to include Asterisk 1.4. :-) -A. ___ --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] 705 DIDs for Collingwood Ontario?
On Thursday 09 August 2007 1:18:17 pm Stephen Bosch wrote: Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) Are there lots of boiler rooms in Collingwood? ... Boiler rooms? (I know what they are, I just don't get the reference...) -A. ___ --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] 705 DIDs for Collingwood Ontario?
On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote: Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) -A. ___ --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 write a function with a return value in Asterisk
On Wednesday 08 August 2007 11:24:45 am Mike wrote: Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. It's been a long-standing sore point with me for sure, since there is no standard way to see if an application returned successfully or not; you have to consult the individual application to see what it sets. -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value inAsterisk
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one specific
purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all done
in the same Asterisk priority (in my case, I want to use the hint
priority but I need to find the value of a variable and use it in the same
line).
Create a function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I can't
know before this line is called (it's very DB driven).
Give the function method a try; that's about the only way I can think of doing
something like that... Note that if it's a very DB driven system, you can
use func_odbc to do what you want by declaring an SQL statement as a
function.
-A.
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Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 12:10:47 pm Mike wrote: I can be a bit slow sometimes, but you said that it's not possible, and on the other hand told me to write my own function (which appears to contradict the first statement). That's because I'm a little slow today... I thought you were asking about writing an application that returned a value. Functions by their very nature return values. As for examples... you've got the source, choose one of the simpler functions and see what you can do. I apologize; I thought you were talking about applications returning values. -A. ___ --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 write a function with a return value in Asterisk
On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote: Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. Digium has taken the stance that Structured Programming is a Bad Idea? I don't think it's fair to paint it quite so broadly. M opinion on it is that I have simply failed to show them how clear things become when I can check ONE variable for the status of the last-run application, whether that be a dial, system or agi application call. Look at the Asterisk source; it's not a mass of spaghetti code. Saying that Digium thinks that structured programming is a bad idea is an exaggeration. -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
exten = 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and you've called it.
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
AGI is not a function. You cannot nest applications like that. The NoOp
application cannot call another application.
-A.
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Re: [asterisk-users] Asterisk Vm functionality question
On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote: Could you elaborate on how you configure the MWI of the mobile device to use asterisk voicemail? yes, please explain. SMSing the phone doesn't light MWI, unless you get access to the raw SMSC, as all the email gateways just mangle the message. -A. ___ --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] Slow list
On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote: Already did that. I use ASSP for filtering. Digium and associated mailing lists are white listed. There was only 1 attempt for deliver and there were no delays. I subscribe to 10 mailing lists (Including the dev list) and they are not having issues. By the way, the only reason I'm able to respond to your messages and I'm watching the archives at lists.digium.com I am having no issues with Digium's lists. They get a little laggy at times, but generally are fast enough. -A. ___ --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] Suing Dell||Dull Computers for CID abuse
On Tuesday 03 July 2007 7:20 am, J. Oquendo wrote: (again) Dell. We know based on someone's accent and lack of proper use of grammar, they are not speaking to us from a location in the USA. How can we validate that such instance is illegal. It You obviously have not been around any city centre in North America if you believe that to be true. :-) -A. ___ --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] Suing Dell||Dull Computers for CID abuse
On Tuesday 03 July 2007 9:47 pm, Joe acquisto wrote: We get to do that, because, back in the late 1700's . . . we won. Hey man, I'm Canadian... We've got our own set of funny accents, and don't get us started on the Quebecois. Not even the Parisians can understand THEM! :-) -A. ___ --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 call quality detection
On Wednesday 06 June 2007 3:33 pm, Jared Smith wrote: Hopefully in the future we'll have the RTCP reports logged (either as part of the CDR records, or in a Call Quality log of some kind). Until then, I'm pretty sure you can listen for RTCP events through the Asterisk Manager Interface, and log them yourself. Yes, you can. The patch is in bug 8613. Unfortunately, the manager events do NOT give you any information which you can link to a particular callno. I've been meaning to add that to the patch, but haven't got to it just yet. -A. ___ --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] wifi sip phone real-world experiences?
On Tuesday 05 June 2007 3:25 am, F6HQZ wrote: I am using Kirk DECT/SIP 600V3 every day. This system run very very well behind an Asterisk, with transfert feature, caller ID display and so... Seen as an IP-Phone running a separate SIP account for each handset. Consider the 600V3 server as a mediagateway converting DECT to SIP. I am Kirk certified (very interesting training at Kirk factory in Denmark) Thank you for posting; tell me, with these Kirk phones, is there any kind of phone book or contact list for speed dials that I can set up? Is it a central list, or does each phone have their own? -A. ___ --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] wifi sip phone real-world experiences?
On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --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] wifi sip phone real-world experiences?
On Monday 04 June 2007 10:28 am, Paul Hayes wrote: Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've got the WIP300 and 330 on my list, with the latter being the more likely candidate, as I can throw up custom apps once I figure out how it's done. :-) I like the idea of a wifi phone running Linux though, so both of these options will have to be investigated. but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. My F1000G phones *CONSTANTLY* lost connection with my WRT54, and it had nothing to do with signal strength, as the access point was less than 10 feet away from my desk, with nothing between to interfere. :-( I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I am also slowly coming to this conclusion. Polycom recently acquired SpectraLink, who've got many years in the wireless phone business. They've got both Wifi and DECT offerings, but nothing with bluetooth, so the search continues. :-) -A. ___ --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] wifi sip phone real-world experiences?
I've tested a few different wifi SIP phones for office/factory use, and generally have been underwhelmed. Before I grab another few and test, I'd like to ask around here about the candidates. My requirements are relatively simple: - WEP/PSK should be supported WITHOUT dragging the phone down - roaming between access points without dropping the call - decent set of ringers, not the garbage that cell phones use now - vibrate, ring, ringvibe (increasing volume a bonus) - transfer, hold, display caller ID - mass deployment (similar to polycom?) TFTP/FTP/HTTP config - decent, but not enormous battery life - replaceable batteries What's not important: - NAT passthrough - colour screen - MIDI ring tones Bonus features: - programmable soft buttons - bluetooth Currently I'm looking at the WIP300/330 and if I can find a source, the CW/Hitachi wifi phone. I've tried some of the UTStarCom phones, Pulver's WiSIP and another I can't think off offhand. Is there anything out there that anyone else has been reasonably happy with? -A. ___ --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] wifi sip phone real-world experiences?
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. -A. ___ --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] Audio going blank for a few seconds andthencomes back. What could be the reason?
On Friday 01 June 2007 9:24 am, Rob Schall wrote: comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would Limited packet loss != **EIGHT SECONDS** of network breakage. Jitter buffers and PLC takes care of most normal network indiscretions, but period dropouts of that big of a time aren't normal and indicate a bigger issue, either with the hardware or the link itself. normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. I have numerous installations using standard telco (Bell Canada and Telus) DSL, and at least one on Rogers cable here in Ontario. No real problems. The odd problem if the pipe gets saturated but careful design and monitoring can take care of most of these problems. I agree with Mr. Hanselman; get a packet logger on the link and see what's really going on. Until that's done, everything here is just speculation. I have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which cause dropouts like this, and I'd like to see what's actually going on before pointing any fingers. -A. ___ --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] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks
On Saturday 26 May 2007 1:21 am, Edgar Guadamuz wrote: Very good... by the way, I'm studing electrical engineering and I've chosen asterisk scalation as my final graduation project. I hope do a similar work within and asterisk cluster. I've been working as an EE, and I've got to ask... what does software scalability have to do with electrical engineering? If you were in a CS prog I could see it, but I've been doing electronic design and power electronics work for over 10 years and I can't think of where these two intersect. -A. ___ --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] meetme sounds
On Thursday 24 May 2007 11:30 am, Steve Edwards wrote: As I remember, the key was to add code to conf_run() to take the user out of the conference, play the custom sound file, and put them back into the conference. These in/out steps are needed to keep that user in sync with the conference. Otherwise, their audio will be offset by the length of the sound file. Eep; I wonder if it would have been easier to mix the sound in to just their copy of the conference... if people were entering or leaving when others were talking it would create holes in the creator's audio. -A. ___ --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] MoH WAY too loud
On Monday 21 May 2007 3:38 pm, Doug Lytle wrote: Doing a 'man sox' does wonders: The question, however, is is Asterisk playing them louder than normal, or are they recorded too loudly to begin with? Adjusting volume gains on these files is the LAST thing you should do. Determine what the nature of the problem is, precisely, before resorting to these hacks. -A. ___ --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] Dry Copper Pair
On Sunday 20 May 2007 11:36 am, Jon Pounder wrote: how many cable feet were you ever able to actually get various speeds at ? Depended on the hardware and wire gauge. I was able to do 1250kbps symmetrical on a 4kmish loop very reliably. around here it might just be the geography but I think load coils are really just a well talked about myth. There are no truly long haul lines due to the number of cities so close together and the lakes blocking what would be any longer haul lines. Load coils are no myth, at least in rural Ontario (Canada) -- I've had to have them removed on more than one occasion. -A. ___ --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 KDE editor to edit Asterisk config files ?
On Wednesday 16 May 2007 11:47 am, Olivier wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? I am a KDE user, although on Slackware. Have been for many, many years. Typically you will find that those who wish to use their GUIs to manipulate Asterisk will do so through one of the available GUIs. Those who want to work on the text files will use vim or emacs. I develop embedded systems; I use kdevelop for coding for the most part, and once in a while I'll use Kate to edit config files, but 99% of my time manipulating text files is done in vim. Even as I type this I have kdevelop open for the source and html, but I have three konsole tabs open: one to a screen session to a server I IRC from, one to a screen session to my development box in the server room (which has two login sessions going), one to a telnet session to the board I'm developing for, and finally one to a serial port server which the serial console of the development box is connected to. Kate's open, but contains a little textfile I append to which has todo lists and notes for the development project. -A. ___ --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] zaptel huge irq problem
On Wednesday 16 May 2007 1:00 pm, François Delawarde wrote: Thanks again for your help, and sorry if I was not 'that' convinced on your first answer and sent a mail to Xen user mailing list to check if they knew that issue (no answer yet). Now I almost believe you a lot. If I understand well I have two options, recode Xen or abandon it. I'll probably go for the 2nd choice and start looking at other solutions, KVM seems to be a good choice and shouldn't interfere much with Asterisk (again: as far as mailing lists say). Let me try to understand this: Xen is a (far) more mature virtualization technology than KVM, and it's been said that it's commercialization was rushed. So you're going to try KVM, which is still under heavy development, as a stable solution? -A. ___ --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] voice recording on legacy PBX
On Wednesday 16 May 2007 1:07 pm, Alex Balashov wrote: You would need two 4-port FXO cards. One to take the 3 outside POTS lines, and one to generate the 3 FXO lines toward the legacy PBX pretending to be the far end. Produce a simple dial plan that basically forwards nearly everything in and out indiscriminately and run MixMonitor() on all of the bridged calls. Uh, you'd need 3 FXS and 3 FXO. You need to generate ring to the legacy system, which requires FXS ports. -A. ___ --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] Feasibility Request
On Tuesday 15 May 2007 3:31 pm, Jeremy Mann wrote: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. I am doing this right now with our MICS. Asterisk is the telco, and routes the calls over our PRI or VOIP provider. I also do a little bit of external extensions. While it works, it's hokey. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. Yes, I was doing this before the PRI. Make sure you're using the right channel bank for FXO, or you won't get CPD. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). Piece of cake, it's just LCR and failover. With the right dialplan nobody knows whether the call went over VOIP or local PSTN. I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. If you want 5 nines out of Asterisk, you're looking at a failover system with a database backend, and T1 failover to the Asterisk boxes. Now you'll also need redundant power and really look at the entire system to make sure there aren't any single points of failure that aren't five nines themselves (i.e. you won't need two PRIs, as they're already considered five nines). Honestly though... take a look at the Citel gateways. Plug all of your Norstar phones into that and connect it to Asterisk. There's your PBX. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). Send me some more information offlist and I'll see what I can do for bidding. Honestly though you'll want to be hands-on on this, as it'll be your butt on the line when (not if) they fall over. -A. ___ --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 obtain the slot position when a call is parked?
On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote:
I want to ask you if asterisk, when I use the command park(), gives me for
example a variable that contains the slot position where it parks the call
or if it only tells me (audio) in the channel this position number? In
other words, is there a way to obtain and use the value of the slot
position when the call is parked? Thanks.
No. You need to use ParkAndAnnounce and a feature I'd managed to get added
which lets you get the parking slot number in the dialplan variable
${PARKEDAT}.
I use it like this:
exten = _X.,1,...
exten = _X.,n,ParkAndAnnounce(PARKED,,Local/[EMAIL PROTECTED])
...
[parkinginfo]
exten = s,1,NoOp(PARKEDAT=${PARKEDAT})
exten = s,n,...
So basically when the call gets parked, it announces the parking slot to a
Local channel which executes in [parkinginfo]. Parkinginfo can write it to a
db, SMS it to a skywriter, whatever you want.
It'd be nice to get this sent to a SIP phone and make parking just that much
more useful, but as the guys say... patches welcome. :-)
-A.
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Re: [asterisk-users] Dry Copper Pair
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote: again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. I'd run about two dozen of these things using a variety of equipment. Pairgain SDSL modems (300S), Flowpoint 2200s, Speedstream something-or-others... hell we even used the flowpoints and speedstreams with an SDSL DSLAM. It works reasonably well in-town, and gets you around a megabit to two, depending on distance. lowest speed I did was about 384kbps, and highest was 2048. All these rates are symmetrical, BTW. In Canada you ask for either a Class A signal channel or a dry pair, depending on whether you are talking to the voice or data guys. You need to get in good with the local tech, too, because if there *ARE* coils, Bell will NOT remove them for you, on the record. The voice circuits have an identifier starting with TVCSNA, and the data circuits CCLADA. These days though, we just order nekkid DSL and get dialtone but no ability to dial anything but 911, and the line's connected to their DSLAM. -A. ___ --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] Dry Copper Pair
On Friday 11 May 2007 7:46 pm, Jon Pounder wrote: well actually there is dialtone on the unprovisioned pairs for the most part, but you can only dial repair, the telco office or 911 on them. I am not sure if its all pairs or just pairs that had a line provisioned at one time. ANAC just replys with some error message if you try to determine the phone number of the line. If you're talking about for DSL use (i.e. connecting to a BAS and using resold DSL service) then yeah, there's almost always dialtone and you can only call the numbers listed. Dry copper (two pairs cross-connected at the CO) has nothing on it. No battery, nothing. Loading coils will be present if one of the loops is exceptionally long, but otherwise it's just as if you'd run the copper between the locations yourself. Where I was located (Listowel, Ontario) we seemed to get better speed vs distance compared to the equipment's ratings, but we chalked that up to having heavier gauge wire in the copper plant (small rural town) and thus less losses in the lines, not to mention possibly a lot fewer competing signals in the trunks. What I am talking about though is if you want to run dsl or some other highspeed type of thing or just an analog pair to a neighbour, or another office in the same neighbourhood, complex etc. All you do is put your tone generator on an empty pair at both locations trace down till you find them in the same F1/F2 box, and jump across them. (no connection to or through the CO, but only possible if both areas are served by the same F1 cable.) Around here at least, the worker who Bell was HIGHLY adverse to this, as it played havoc with their planning, at least according to them. We were only able to have them cross-connected at a pedestal in one (early on) loop; all others were REQUIRED to run through the CO, which often added too much distance for us to make it useful. F2 pairs, or even noticed it and cared, who would know ? Actually it would probably take some investigation to even tell if its a legitimate bridge tap or the left overs of one or just something that is not supposed to be there at all. In a world of if its not broke don't touch it, it would likely never get touched. That's the other thing we ran into from time to time: bridge taps. Loops that should have gotten an easy 1.5meg wouldn't sync at all, and eventually the culprit was found to be a 5km tap run off to some new subdivision. -A. ___ --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] Re: RE: Digital Phones
On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote: I understand that these sets are digital but what about connecting Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel? Yes you can do that; I have. No you don't want to; it doesn't work worth a shit. You lose so many features, you are constantly putzing around with it, and it never works as good as you'll hope. -A. ___ --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] Channel Bank
On Sunday 06 May 2007 6:42 pm, Forum wrote: Can someone recommend a good quality 24 or greater port channel bank? For FXS: I have personally used Adit600, Access Bank I and IIs. They all work great, and the AB1 and AB2 products are *cheap*. For FXO: Adit600. The AB1/2 work, but have no CPD capability, so you can never tell when the other side hangs up on you. Rhino makes 'em too, as does Adtran, but I have no experience with these. -A. ___ --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] Semi-OT: useful things to do with XML browsers in phones
On Thursday 03 May 2007 10:18 am, Chris Bagnall wrote: It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? Queue stats, sales/donation volumes, weather/stock/news, door/gate alarms... -A. ___ --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] delay in switching between contexts
On Wednesday 02 May 2007 11:49 am, Danish Samad wrote:
[salesivr]
exten = _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id
${CALLERID})
exten = _X.,2,Playback(emptyy)
exten = _X.,3,Background(Main_Sales)
exten = _X.,4,WaitExten(2)
When I press a digit in _X,3 or _X,4 it takes some time before switching to
the desired context. When switching context I see this on the cli
This is because you are telling asterisk to wait for one OR MORE digits, and
it is waiting to see if you're going to dial any more digits before deciding
that the entire extension has been entered.
Use the various timeout functions to set a more appropriate timeout, or code
your dialplan to only look for one digit. i.e. exten = _X (no trailing .
(period)).
exten = _X.,5,Goto(_X.,3)
This works, but looks very odd to me. Just specify the priority if you want
to stay in the same extension... i.e. Goto(3).
-A.
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Re: [asterisk-users] allowing call every 15mins
On Wednesday 02 May 2007 3:04 pm, Goke Aruna wrote:
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is 15, 30, 45, 60(in multiples of 15 minutes).
So YOU're the guy who makes the calls to tech support so hideous!!
how can i achieve this and what application can i use to get this done.
GotoIfTime can help you here, but it'll be a little messy:
exten = 1,1,GotoIfTime(0:01-0:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:16-0:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:31-0:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:46-0:59|*|*|*?toobad)
exten = 1,1,GotoIfTime(1:01-1:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:16-1:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:31-1:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:46-1:59|*|*|*?toobad)
...
exten = 1,1,GotoIfTime(23:01-23:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:16-23:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:31-23:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:46-23:59|*|*|*?toobad)
exten = 1,n,Dial(SIP/techsupport)
exten = 1,n,GotoIf($[${DIALSTATUS} = BUSY]?toobad)
exten = 1,n,Hangup
exten = 1,n(toobad),VoiceMail([EMAIL PROTECTED])
Very messy. Alternatively:
exten = 1,1,GotoIfTime(0:00-0:00|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:15-0:15|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:30-0:30|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:45-0:45|*|*|*?woohoo)
...
exten = 1,n,GotoIfTime(23:00-23:00|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:15-23:15|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:30-23:30|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:45-23:45|*|*|*?woohoo)
exten = 1,n,VoiceMail([EMAIL PROTECTED])
exten = 1,n,Hangup
exten = 1,n(woohoo),Dial(SIP/techsupport)
...
Pretty much equally messy.
Both of these examples assume you want to allow calls for a minute every
quarter hour 24 hours a day, quite possibly to match policies on most vendors
which claim they offer 24-hour tech support but implement similar
dialplans. :-)
Honestly though this is a strange request... Why bother offering tech support
if you are only allowing calls for 1 minute every 15 minutes? Why not be
honest about it and do this:
exten = 1,1,Playback(sorry-we-dont-offer-support)
exten = 1,n,Wait(30)
exten = 1,n,Hangup
??
-A.
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Re: [asterisk-users] ADSL routers with integrated SIP QoS for other devices
On Monday 30 April 2007 4:14 pm, bails wrote: I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Any chance we can get to see this as it sounds just what i'm looking for? Once I find something, yes. :-) If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. Forbidden You don't have permission to access /~andrew/rc.tc on this server. Fixed. -A. ___ --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] ADSL routers with integrated SIP QoS for other devices
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. -A. ___ --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 Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. -A. ___ --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 Planning Help
On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote: I am trying to setup an arrangement whereby clients on machines A, B, C and D can talk to each other on Softphones. A,B,C are are all Windows XP machines, machines D and S are linux. This has to include A talking to B and ultimately conference calls with potentially all parties. Personally I make my Asterisk box the firewall. It eliminates all NAT troubles. :-) If that's not your style, I'd use IAX over SIP, as it only requires a port-forward to D on D's NAT box. SIP you may be able to get work with port forwarding 5060 and 1-2 (all udp) over to D, but I'm not sure... Naturally, nat=yes and canreinvite=no should be set all around. -A. ___ --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 Faxing Support
On Wednesday 28 February 2007 7:53 pm, Lee Howard wrote: The problem, however, as we all know, is that the Asterisk maintainer, Digium, requires undue retribution in the form of disclaimers before it will accept any contribution into the code repository - and in this case the author of the desired contribution is reasonably refusing. Undue? Digium requires disclaimers so they can dual-license it for ABE and other commercial vendors. You're purposely twisting and distorting the reality with these weasel words. If you don't like it, use something else. There's no need to take jabs at the company. -A. ___ --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] To use asterisk or proprietary hardware, that is the question
On Saturday 24 February 2007 6:48 pm, Matt wrote: Now.. back to your issue. Setup a crontab to restart asterisk every night. Use a version of Nonsense. Set up proper monitoring of system resources (memory is only one resource you should be watching) and help the community out if you're detecting memory leaks. restarting every night is bad bad bad. asterisk you know well (I like 1.2.6) and know is stable.Finally, setup RAID-5 on the hard drives. That way if one dies, you can still replace it Again, nonsense. software RAID1 is more than adequate, but personally I far prefer to use CompactFlash. There's absolutely no reason to have three+ drives in small office PBX; Hell I'd be hard-pressed to justify two (RAID1) in such an install. -A. ___ --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] b410p + fax (echo cancellation)
On Friday 23 February 2007 8:35 pm, Zoilo Gomez wrote: However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. Generally speaking all modems (this includes POS machines and faxes) emit a tone which echo cancellers recognize and disable themselves for that call. -A. ___ --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] The High Performance Echo Canceller (HPEC)
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote: It looks like octasic have started supplying their echo canceller as host software for zaptel now. I expect either canceller would work with the Sangoma cards, as they currently sit in the zaptel framework too. Out of curiosity, why do you suppose that it is the Octasic algorithm which is used in Digium's HPEC? I have no reasons to suspect otherwise, but I'm curious as to your reasons for suspecting that is indeed the case. Oh, and sorry about the incorrect attribution as to which Steve wrote and maintains spandsp. I always get yourself and Steven Critchfield mixed up. :-) -A. ___ --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] End Wrap-up Time?
On Tuesday 13 February 2007 11:30 am, James Fromm wrote: Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... -A. ___ --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] End Wrap-up Time?
On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --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] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 11:17 am, shadowym wrote: I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. Why not? A DSP is a CPU which has been designed to do mathematical functions very quickly, generally especially with respect to matrix math. I mean think of what you just said. You could just as easily have said A CPU ... it's an ASIC. Everything it does is completely in hardware. -A. ___ --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] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 11:19 am, Matthew Fredrickson wrote: We noticed that it has slightly better performance characteristics than the Octasic, particularly in double talk scenarios, at least from our internal lab testing. How has the testing been with respect to its use on FXO ports (such as those on the TDM400 FXO modules) ?? I'm *very* interested in any real test data, including any comparisons with MG2 and the Octasic cancellers available on Digium products. -A. ___ --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] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 4:12 pm, shadowym wrote: The algorithms may be similar but EC is an infinitely variable non-linear(analog) process. A CPU cannot do that. You can fake it by performing cpu intensive rapid calculations one after another but it is fundamentally not an analog processor. HWEC is designed to deal with the analog process on an instant by instant basis performing parallel computations. A CPU cannot do that at ANY clock speed. I think you are very sorely mistaken. I've done DSP work on general-purpose CPUs for many years. All current processors have SIMD, which, until the i586 (for Intel), was more or less only in DSPs. Steven Critchfield has been doing DSP work (spandsp) for much longer than I have, and is much better at it than I will ever be. :-) Anything a DSP can do, a general-purpose CPU can do, but very likely slower. There is no magic. There is nothing particularly special about ASICs or DSPs that general-purpose CPUs can't do; it's all a matter of how quickly it can do it and how much you're willing to consume in system resources. -A. ___ --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
