Re: [asterisk-users] A reason TO run Asterisk as root
tune2fs -m 0 [device] :) not anymore ;p David Backeberg wrote: On Wed, Jul 22, 2009 at 11:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Hehe, sounds like a reason to standardize on ReiserFS ___ -- 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 -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] MeetMe not prompting for PIN
No That says: Join Conference 123456 The PIN for the Conference is 123456 What you need to do is setup the conference ID, guest PIN, admin PIN in the meetme.conf and then use exten = ,1,MeetMe(123456,cMaAsxp,) John A. Sullivan III wrote: Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten = ,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I have tried it both ways and tried using the P flag. The user is never prompted. What am I missing? Thanks - John -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] IP phone recommendation
I'll put in my $0.02CAD for Polycom. We use the 330s here. Singer Geraint Lee wrote: I personally find the snom phones to be generally ugly and un-finger-friendly, in terms of reliability and quality, never had any trouble, good phones all in all, i just can't get past the tacky look and feel so don't buy them. 2009/6/3 Darrick Hartman dhart...@djhsolutions.com mailto:dhart...@djhsolutions.com On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote: On Thu, 4 Jun 2009, Rob Hillis wrote: Jeff LaCoursiere wrote: We are still talking about a $175 phone. How about the Polycom IP 320? $85 at 888voipstore. Can't go wrong with Polycom for voice quality. True, Polycom's are brilliant for voice quality, but unlike the Snom, a Polycom /will/ reboot on the drop of a hat /and/ take a damned long time to do it (~45-60 seconds) In addition, the web interface should be taken away and shot - the only real way to configure them is through (T)FTP. They are however, extraordinarily configurable through the XML config and they are very stable. Once they're configured they work very nicely. The lack of a decent number of BLF keys (even with a very expensive sidecar you only get two more keys than a standalone Snom320) puts me off a little. However, for a conference phone, the Polycom's can't be easily beaten. Their handsfree call quality is in a league of it's own. Mainly I suggest it because the OP asked for an inexpensive quality phone. I agree on the provisioning - the web interface is useless, and unless you know how to setup the XML files properly you are doomed to a very frustrating experience. The Polycom 320/330's are nice little phones for the price. There are several resources for configuring the phones from the XML config files. If the config files are sane, the phones don't take that long to reboot. This is probably one of the better examples: http://www.kfife.com/voip/ Karl did a good job commenting in the config files where he made changes. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] VoIP over satellite internet
David Gibbons wrote: snip ...routing via satellite adds about a quarter second of latency to the path. Is that too much? /snip Eric, I believe that you are mistaken. Routing via satellite adds about a quarter second of latency PER TRIP from earth to orbit. This is simply due to the distance a satellite is from the ground and the speed of light (interference not withstanding). Traceroutes and pings to satellite providers can be misleading because they cache some content on the birds in order to decrease latency. As I recall they even intercept some pings to accomplish the same. A *real* round trip for a VOIP call and/or non-interfered TCP connection would look like this: 1. Your device up to the bird (~250ms) 2. The bird back to the ground (~250ms) 3. The ground station out to the internet (~Nms) 4. The internet back to the ground station (~Nms) 5. The ground station back to the bird (~250ms) 6. The bird back to your device (~250ms) As you can see, even the one way udp stream will take approximately 500ms beyond any latency introduced by things such as your wireless network and the internet. VOIP over satellite, as Josh indicated, will be painful. You'll be talking all over one another due to the delay assuming that the stream can even be sustained with that much latency. -Dave ___ -- 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 Of course, that's assuming your satellite is in geosynchronous orbit. If its in LEO, then its much better. Singer ___ -- 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] Please Advice SIP 183 progessl
What's wrong with top posting?
David Gibbons wrote:
Lol.
I'm actually in the small minority who prefers top posting to bottom posting.
-d
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, April 03, 2009 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
And don't top post ;)
On 3 Apr 2009, at 14:38, David Gibbons wrote:
The fact that you sent this again (what is that -- 3 times now?) AND
with high importance, will likely cause people to ignore your
messages rather than trying to help you.
There are few things that annoy me more than messages sent with high
importance (same category of annoyance as messages written in all
caps). Let's have a little bit of intarweb etiquette.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
] On Behalf Of Khaled W. Chehab
Sent: Friday, April 03, 2009 9:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Importance: High
Dears
Kindly find my dial script below,I am trying to send the caller 180
ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
or if there any way to stop the music on hold and let the caller
hear the
Ring Back Tone
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no
answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my
source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
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Re: [asterisk-users] Forking
At least fake your from email to make it believable.. hh174 wrote: Hello all, Probably a bad news for all... The Undercompetent Olle E Johansson decided to leave the asterisk team to create his own Voip server. The server will be called Minisk (due probably to his poor competence in Voip). Following that, Digium decides to stop any development on Asterisk and joined the Skype team to recreate a brand new paying licensed software. Any development will be halted for asterisk It seems that Mr Johansson has was poisoned by a fish in Brussels during the last Fosdem. We all hope qur Mr Johansson will quickly heal and return to the team of asterisk. Kind regards, Olivier Taylor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to find small footprint asterisk platform
Tzafrir Cohen wrote: On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote: One of the more common embedded platforms for Asterisk is the Soekris net5501 (or 4501 if you don't need as much processing power) Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for almost the same money (Soekris stuff isn't cheap in the UK) and is about the same footprint, it might be worth considering that instead if you don't need ISDN or POTS connectivity. I've done a few Asterisk-based eeeBoxes over the last few weeks and been very impressed with them. In fact, with a netbook I suspect you'd be paying quite a sum for the display. Both in the price and in the heat consumption. I'm confused, what display? -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Need to find small footprint asterisk platform
Sorry, forgot a link http://www.eeextra.com/eee/eeebox-specs.html Singer XJ Wang wrote: Tzafrir Cohen wrote: On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote: One of the more common embedded platforms for Asterisk is the Soekris net5501 (or 4501 if you don't need as much processing power) Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for almost the same money (Soekris stuff isn't cheap in the UK) and is about the same footprint, it might be worth considering that instead if you don't need ISDN or POTS connectivity. I've done a few Asterisk-based eeeBoxes over the last few weeks and been very impressed with them. In fact, with a netbook I suspect you'd be paying quite a sum for the display. Both in the price and in the heat consumption. I'm confused, what display? -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Hardware suggestions
Mike wrote: You can reliably run asterisk on just about any x86 hardware. You don't mention what kind of stresses you are going to put on it, so your sizing questions are impossible to answer. How many extensions? How many simultaneous calls? Will you be transcoding? Routing to/from the PSTN? What cards will you be putting in the box? Some cards don't play nicely together if forced to share interrupts, for example. Sizing is important. Take your company's projected growth rate, double it, and work it out for 3-5 years. I recommend 5 years for the sizing. As much as its fun to tinker, once it goes into production you want to have it as stable as possible. Look at all the apps you want to use and figure out how much they are going to cost you in terms of resources. In the company I work for, we put in Asterisk to replace our Nortel system which reached the limits. So we expected standard usage rates and growth etc. However, once we introduced meetme application our Asterisk usage spiked. We figured on average 2-3 meetme meetings a week (based on the usage of a third party conference bridge we had before), and now its at 2-3 a day. We had it setup so that every person has their own conference bridge. Other features are also taking up more resources. I'm currently modigying meetime and writing an AGI so that once the meetme conference ends, it will take the recording and conver it to an mp3 and then emails it to the leader. I wasn't worried about sizing (let's imagine that this is more than enough for now and less than I'll need later). More about whether this was the right BRAND more than the right hardware. Does HP make Asterisk friendly hardware? I know Dells was problems a few years back. As for CPU, the question is mostly one about more GHz or more cores? Dual cores are cheaper by GHz. What`s best for Asterisk? I am doing only SIP to SIP calls. Some transcoding (half calls are G711 to G729, the other half are G729 both ways). [snip] I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
Mmm, $100-$125 What? USD? CAD? AUD? If you're willing to a little bit more, I'll strongly recommend Polycom IP 430. We're using them and they are absolutely painless (well, except the initial package of 100 of those which were heavy and caused some back pain ;p) Singer David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Quiet 24 port POE gig switch
[snipped] You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. How are you getting these 80 or 120mm fans in a 1U chassis? Remember you got barely 45mm to play with at the back and front of the switch. How are you going to mount a 80mm or 120mm fan on there? Are you assuming that the units mounted above (or below) your switch is a short 1U? You can't assume that... begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Quiet 24 port POE gig switch
We got a few of those in 1U chassis.. if you think those are quiet... Steve Underwood wrote: Singer XJ Wang wrote: [snipped] You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. How are you getting these 80 or 120mm fans in a 1U chassis? Remember you got barely 45mm to play with at the back and front of the switch. How are you going to mount a 80mm or 120mm fan on there? Are you assuming that the units mounted above (or below) your switch is a short 1U? You can't assume that... A number of 1U products use large impeller fans. They take more internal space in the chassis, but they are slower and quieter. Tiny whiny seems to be the current fashion, though. Steve begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Quiet 24 port POE gig switch
Honestly, how are you guys expecting a 24 Port POE to be fanless? Lets start with some logical points here: 1) 24 Ports x 15.4W/Port = 369.4Watts + Switch Power = ~400Watts... now Power Supply isn't that efficient so you're getting probably a 500Watt Power Supply (assuming 80%)... 2) with a 1U chassis, you can't blow air up or down... only front and back.. so you're stuck with a 40mm fan.. ... Benny Amorsen wrote: Paul Hales pdha...@optusnet.com.au writes: My memory of a HP procurve (a 2626 PWR from memory) was that it was quite noisy - have they changed? The 2626 is either extremely noisy or fairly noisy, depending on which you happen to get. Luck of the draw; I haven't found a way to predict it. The 2650 is almost always in the fairly category. However, the 2610-24 is fanless, which sounds great. Except the PoE versions aren't fanless. The 2610-24/12PWR is not THAT bad unless it gets hot, the 24PWR and 48PWR are fairly noisy. /Benny ___ -- 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 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Quiet 24 port POE gig switch
Okay, point out one reasonably priced PoE switch that has it. Christian Victor wrote: 2009/2/2 Singer XJ Wang w...@pythian.com mailto:w...@pythian.com [snipped] You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. How are you getting these 80 or 120mm fans in a 1U chassis? Remember you got barely 45mm to play with at the back and front of the switch. How are you going to mount a 80mm or 120mm fan on there? Are you assuming that the units mounted above (or below) your switch is a short 1U? You can't assume that... Ever heared of a centrifugal fan? ;-) Chris begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] [Asus Eee PC 900] as replacement for legacy BRI phone
As much as I'm an open source guy, but the OpenMoko phones are worthless IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it too much to ask for at least EDGE if not 3G? Matthias Apitz wrote: El día Wednesday, January 07, 2009 a las 09:50:06AM +0100, Grygoriy Dobrovolskyy escribió: Xorcom had something, usb bri, but it is pricey. If you dont need to change provider and planning to stay with bri, why dont you buy another bri phone ? Because since I own the Moko I know *what* it means to have a UNIX based (cell-) phone: You can SSH to it and change/install/write-by-your-own whatever you want or what you feel missing. Now I want to have a UNIX based bri phone. That's it, not more, but not less :-) matthias -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] [Asus Eee PC 900] as replacement for legacy BRI phone
As much as I'm an open source guy, I'm also big on making my life easier. I use to bottom post till I got a BlackBerry. With BB (and also Treo, Windows Mobile, iPhones) its just easier to read with top posting. The phones generally download first X (I belive X = 4 for the BlackBerry) KB of each message. Matthias Apitz wrote: El día Wednesday, January 07, 2009 a las 10:55:55AM -0500, Singer XJ Wang escribió: As much as I'm an open source guy, but the OpenMoko phones are worthless IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it too much to ask for at least EDGE if not 3G? We go offtopic of the list, I think; one last statement from me: the OpenMoko is the best phone you can get today (I'm using it as my daily phone); and it is the only one you can SSH to and install/write what *you* want; I don't need more then GRPS for data (I'm an ASCII guy for e-mail and USENET); and I have GPS, GSM, full Linux, bash, X11, GTk, Python ... really, there is no better option for open source folks at the moment; matthias PD: if you are an open source guy, you should not top-post -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] 5 min limitation on phone calls! how to!
Phone Guy: NO PHONE FOR YOU!
Karl Fife wrote:
This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl
On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
i tried that before.. it didnt actually work! it the call kept on going
well beyound the allowed test seconds...
heres my extensions.conf:
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Thu, 21 Aug 2008 20:26:48 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
RoLaNd RoLaNd schrieb:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for this!
Set(TIMEOUT(absolute)=seconds)
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
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Re: [asterisk-users] 5 min limitation on phone calls! how to!
Heck, I was going to say I probably be on the sofa that night and the
next...
[EMAIL PROTECTED] wrote:
You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
From: Singer XJ Wang [EMAIL PROTECTED]
Date: Thu, August 21, 2008 2:42 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Phone Guy: NO PHONE FOR YOU!
Karl Fife wrote:
This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl
On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
i tried that before.. it didnt actually work! it the call kept on going
well beyound the allowed test seconds...
heres my extensions.conf:
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Thu, 21 Aug 2008 20:26:48 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
RoLaNd RoLaNd schrieb:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for this!
Set(TIMEOUT(absolute)=seconds)
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
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http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE
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