Re: [Asterisk-Users] RE: Sangoma VS. Digium
cmould wrote: Where is this discussion going. I am about to do an installation that will require t1 interfaces. I am new to the telephone world and found the original discussion useful. I need to know from a reliability and performance standpoint what is the better choice. Sangoma or Digium? Sangoma cards are waaay stable. I've traded my Digium TE410P to Sangoma A104 card, and lots of troubles a had is gone. Before that I had a lots of HDLC errors ( even with very small channel load ) which is caused the random disconnects of my customers. For now I didn't see any FCS HDLC error for quite long time. Even Sangoma cards works in crippled mode ( and loosing as minimum as 100 times of the performance ) it's still have better performance. This is my opinion. I am not demanding to agree or disagree to me. Everyone makes their own choice. I made mine. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext 37 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Liveviop problem
tinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. ========= Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext 37 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium : no lead time!
If you are in GTA, and willing to drive to Markham, you can have the Sangoma card at the same day, even in few hours =) PS: Okay I understand this is place only for Digium cards. But I traded my Digium TE410P card, and bought Sangoma card few hours later directly from manufacturer and I am quite happy with that. Don Murray wrote: sales testimonial on I ordered a TE110P card from Digium on-line on Monday. I got email confirmation and FedEx tracking number the same day. I was sick on Tuesday. Wednesday morning the card is sitting on my desk when I come in to work. And I'm in Canada too, so it had to cross a border. Thats pretty fast service! I didn't even request the express shipping option. Meanwhile, the local reseller who I emailed on Friday to ask regarding availability hasn't even returned my email yet ;) sales testimonial off Don ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext 37 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice-like company in Canada?
Their international rates are very high. We are planning to provide North American plan in very short time, but our rates for SIP connections are wy lower. You can check it on our web-site: http://www.hitcalls.com If you want to be connected via SIP, just drop me a few lines to my email. We can even configure IAX2 connection for your * server, and provide a local 416 area code number. This is still in beta stage, but works very well. We have a beta-testers for this service, and some of them even posting in this mailing list. PS: Probably this is wrong list for some ads ( it's suppose to be in asterisk-biz), but if people looking for good rates, this is our pleasure to help. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext 37 JR wrote: Mobitus seems very cool. Here is another question for the group: Is it easy to use Vonage with an * box? Could I order service from them and not use the equip. they send? -JR On 8-Mar-05, at 9:34 PM, jurgen wrote: Hi Justin, I used to work with the fine people at Mobitus. (www.mobitus.com). Give them a try. Last I looked, they have some kind of free trial offer. ...jurgen On Tue, 8 Mar 2005 20:38:13 -0500, JR [EMAIL PROTECTED] wrote: Hey folks, I am looking for a no frills bring-your-own-SIP device VOIP company similar to Broadvoice. Does anyone have any experience with VOIP in Canada? -JR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] storing cdr in two databases
I don't know how about mysql, but cdr_pgsql.so works exactly that way. I do have and files, and records in the Postgres. Kevin P. Fleming wrote: Ludovic Drolez wrote: Is it possible to send CDR to a database (cdr_mysql.so for example) and to files (cdr_csv.so) ? Not currently, no. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
Easy as piece of cake. Remove ignorepat=9 add: exten = 9,1,DISA(no-password|my_outbound_context) [my_outbound_context] exten = NXX, 1, blah-blah-blah All the Best! Sergey. Peter Svensson wrote: On Sun, 20 Feb 2005, Anton Krall wrote: Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? I'm not sure I understand your question. Do you want to be able to hit 9 and get a an outside line with dialtone? Just add an extension to do that. For isdn you need to enable overlap dialing. Or do you want Asterisk to provide a dialtone after the user have hit 9 as the first digit of a number? User the ignorepat option in the dialplan. Or do you want Asterisk to provide a _different_ dialtone after the user have hit 9 as the first digit of a number? This may be possible, but I think some hack may be needed. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
Yesterday, I've checked tariffs from Bell Canada, For Full voice T1 it was costs around $1000 + tax. $216 - is access fee, $34 per channel. You can get the PRIs from Allstream with 3 years commitment ~$600 per month. Andrew Kohlsmith wrote: On February 20, 2005 11:44 am, Jim Van Meggelen wrote: I like the thinking; the challenge is often where in the world you are, and how much competition there is. Here in Ontario, T1's were generally priced such that fractional T1s hardly saved anything. There is more competition now, so prices are changing, but I still can't see frac T1 service competing with such a small number of analog circuits. I know there are places where such a thing could be had very competitively, so your advice is still good. I think you'd be surprised. Even in Listowel a CT1 for POTS termination was on-par with having the individual analogue lines brought out. You'll pay a little more for the smartjack lease but it eliminates a lot of headaches. Hell the PRI here in cow-town Listowel was in-line with POTS until you included the D channel price of $500 -- The B chans were all $55/mo which is exactly what a business line costs. I imagine CT1 instead of PRI service would have been significantly cheaper, *AND* I wouldn't have to pay for all those extra DIDs. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort
This is happens because of imperfect HDLC code. I am having the same in my logs, but quite rare and on spans which is idle. Therefore this is not an issue with PRIs itself. I may be wrong, but telco technician checked my PRIs as well, and didn't find any flaws. I can tell you more. IT happens when server not being rebooted for quite long time. Right now I am rebooting server every day or two. Alex G Robertson wrote: Some news. It is not caused by transmission lines, conectors or anything like that. The telco tecnician just came here and analyzed the circuit and he got no erros! He sugested me to loop my PRI port in the balum attached in my asterisk box. And Surprise... I got the same errors! The error is on my hardware/software. []s Alex Robertson Alex G Robertson wrote: Hi everybody, I just installed asterisk, but this NOTICE dont stop appearing on my log file;; Feb 17 18:30:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:29:42 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:29:41 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:29:41 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:27:11 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:26:51 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:25:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:24:41 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:22:21 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:16 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:13 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:12 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:01 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 And from time to time this is happening -- B-channel 0/1 successfully restarted on span 4 -- B-channel 0/2 successfully restarted on span 4 -- B-channel 0/3 successfully restarted on span 4 -- B-channel 0/4 successfully restarted on span 4 [...] -- B-channel 0/29 successfully restarted on span 4 -- B-channel 0/30 successfully restarted on span 4 -- B-channel 0/31 successfully restarted on span 4 And the conversation stops. Telco, with a traffic analyzer, says that the clock is sliding. Does anybody knows what can it be? Hardware, software, transmission (conectors) etc ? Thanks in advance. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wiki down?
Or I can host it. I have a few colo servers at 1 and 1 with half of terrabyte of monthly traffic on it. Multiple connections to Tier-1 providers, 12 Gbit total bandwidth. I would host it with pleasure. Nir Simionovich wrote: Well, I have no idea where the wiki is hosted, but if the wiki needs to be moved to a more stable location, our hosting facility in Israel is as stable as you can get. We have 2 circuit running in, BGP4 and an uplink of 4Mbps. I'm confident it should be enough, no? Nir S - Original Message - From: Sascha E. Pollok [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 19, 2005 9:36 PM Subject: RE: [Asterisk-Users] wiki down? Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk Sendt: 19. februar 2005 19:14 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] wiki down? hi is the wiki down again? roy Ain't there any mirrors available? Hm... might be difficult with the database/dynamic content at the back end... Else I might be willing to host one ... Cheers Sascha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have a odd question...
Easily. AGI script + DB. [EMAIL PROTECTED] wrote: Hi all. I am going to do a simple voting application for a radiostation. The idea is to have listeners call in to vote on songs. What I want to do is to take a phonenumer for each song and present the result on a simple webpage. Eg. To vote on song number one, call 555- To vote on song number two, call 555- etc etc. When the listener calls in, a playback tells him: Thank you for voting on song number one. And the numbers of calls on each number are presented on a webpage, or in a textfile, easy for the showhost to see. How do I do this the simplest way ? I have a lot on phonenumbers that I can use, so that is not the problem. Shoud I execute some kind of script for each caller that increases the numbers in a textfile ? Or how should I do ? My programmingskills aren't the best, so I would be greatful for any help I can get. /Regards Mike. PS. Please answer offlist if possible.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2: Connection rejected
Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/user/1' Even I have entry in iax.conf for this user as a friend, and * server of this user is already registered with my * server. I can't register with his box because: 1. his IP is semi-dynamic. 2. this is nonsense - His box already registered with mine. Is there any solution? Thanks a lot in advance! -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 x 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2: Connection rejected
They are the same. That's what I've checked first. Peter Bowyer wrote: On Wed, 16 Feb 2005 15:40:19 -0500, Sergey Kuznetsov [EMAIL PROTECTED] wrote: Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found Is there any solution? The log is telling you that the remote server is refusing the connection from your server because of incorrect authentication. Check the IAX peer/friend entry in the remote server against the credentials you're using in your friend entry or in the dial string. Peter -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2: Connection rejected
Thats what I already have. Here is the entry: [user] type=friend accountcode=XX amaflags=billing host=dynamic secret=mostsecret auth=md5,plaintext context=iax_out disallow=all allow=gsm allow=ulaw allow=alaw allow=adpcm callerid="User" 416XXX trunk=no jitterbuffer=yes dropcount=5 tod=lowdelay the user's iax.conf has: register = user:[EMAIL PROTECTED] [myserver.com] type=friend host=voip.myserver.com secret=mostsecret trunk=no context=default auth=md5,plaintext callerid=416XXX Steve Totaro wrote: not a permanent solution according to many on the list but try type=friend in your iax.conf - Original Message - From: "Sergey Kuznetsov" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 3:40 PM Subject: [Asterisk-Users] IAX2: Connection rejected Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/user/1' Even I have entry in iax.conf for this user as a friend, and * server of this user is already registered with my * server. I can't register with his box because: 1. his IP is semi-dynamic. 2. this is nonsense - His box already registered with mine. Is there any solution? Thanks a lot in advance! -- All the Best! Sergey. ===== Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 x 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. ===== Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec for X-lite soft phone
Adrian, Have you ever read the note for that? =head Comment * NOTE G.729a for X-PRO Pocket PC is available with a minimum order of 20,000 units, G.729a for X-PRO Mac OSX is available with a minimum order of 10,000 units, G.729a for LindowsOS is not available at this time. =cut Adrian Chapman wrote: Daniel Eboa wrote: Sir, I think when somebody asked a question, is because he doesn't know the answer. Even maybe when for some people like you, the answer is evidence. Thinking that I know the answer of the question I asked, suppose that I'm stupid, while I'm not. I you feel offence by the question I asked, please simply ignore it. Regards. Daniel. Daniel, I think you'll find that Seshu was implying you do a little basic research for yourself. It's easier than asking here, and you get an answer more quickly. A two second visit to www.xten.com, and clicking on the clearly labelled link to compare the free and paid versions of the softphone shows that one difference between the free and paid versions is that the paid version supports G.729a. Get your credit card out, and your desires are met. http://www.xten.com/index.php?menu=productssmenu=compare -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec for X-lite soft phone
There is the red asterisk near of G.729a which is footnotes to this note. Adrian Chapman wrote: Sergey Kuznetsov wrote: Adrian, Have you ever read the note for that? =head Comment * NOTE G.729a for X-PRO Pocket PC is available with a minimum order of 20,000 units, G.729a for X-PRO Mac OSX is available with a minimum order of 10,000 units, G.729a for LindowsOS is not available at this time. =cut It reads as included with a minimum order of 1 unit for Windoze, though. I may be wrong. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec for X-lite soft phone
Oops, my fault! Disclaimer: Where is my favorite coffee mug ?! Dana Olson wrote: And for Windows, a minimum purchase of 1 unit... Is he using Mac or PocketPC? If not, then he doesn't have to worry. On Wed, 09 Feb 2005 15:59:45 -0500, Sergey Kuznetsov [EMAIL PROTECTED] wrote: Adrian, Have you ever read the note for that? =head Comment * NOTE G.729a for X-PRO Pocket PC is available with a minimum order of 20,000 units, G.729a for X-PRO Mac OSX is available with a minimum order of 10,000 units, G.729a for LindowsOS is not available at this time. =cut Adrian Chapman wrote: Daniel Eboa wrote: Sir, I think when somebody asked a question, is because he doesn't know the answer. Even maybe when for some people like you, the answer is evidence. Thinking that I know the answer of the question I asked, suppose that I'm stupid, while I'm not. I you feel offence by the question I asked, please simply ignore it. Regards. Daniel. Daniel, I think you'll find that Seshu was implying you do a little basic research for yourself. It's easier than asking here, and you get an answer more quickly. A two second visit to www.xten.com, and clicking on the clearly labelled link to compare the free and paid versions of the softphone shows that one difference between the free and paid versions is that the paid version supports G.729a. Get your credit card out, and your desires are met. http://www.xten.com/index.php?menu=productssmenu=compare -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please share the experience on VoIP phones heavy using.
Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please share the experience on VoIP phones heavy using.
Jerry, Thanks a lot for the feedback! By the way, how long did it take to replace the faulty 10% of phones by RMA? What company did you use to buy it from? Jerry wrote: On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote: Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. Both have excellant sound. I think the Polycom speakerphone is a bit better. We are using mostly Polycom these days and our customers love them. My only issue is they do seem to have about a 10% failure rate within 90 days. After that they are solid - so far. They are also less expensive than the Cisco's and seem to have a better feature set and better control of their configs and buttons. I do like the layer 2 troubleshooting capabilities of the Ciscos as the Polycom seem to have no capabilities that I can find. I do not think the Cisco is any kind of a Polycom copy. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 ext. 37 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Criteria
I have Dual Opteron 2 GHz with 4 Gb memory. I don't have huge load right now, and system load is almost 0.00 even if it uses slinear to G.729 transcoding. I have Wildcard 410P installed. Works good for me. Spencer Nassar wrote: I've been doing a lot of background reading/searching of this list, voip-info.org, and Google, looking to define a good candidate for a server platform. I'm very interested in thoughts from others! So here goes... Axiom 1: if you are not doing doing much transcoding (converting between codecs), the bottleneck for supporting high volumes of simultaneous calls is system bus speed, not CPU power --- points to a 64 bit AMD Opteron system, and maybe just one of the two processor slots populated. Bus is twice as wide as a 32 bit system, and operates at 1.8GHz (a lot faster than a 64 bit Zeon system). Then add the second processor to the board if you see you need it. Axiom 2: Get lots of memory --- I haven't seen this quantified, and plan to do some testing. I'll post results here, but can anyone share any insights? I'm planning to start at 2GB, and go up from there if I see swap getting used. - what would an alaw to alaw connection consume (if it didn't hand off)? - what about a 5 call alaw meetme bridge (and how much memory per incremental caller) Axiom 3: Don't allow any disk IO --- I'm assuming this is related to #2 - get lots of memory to avoid swap to disk. Other issues or thoughts? Axoim 4: Come codecs will take advantage of the faster floating point of a 64 bit system --- unknown... has anyone seen this? Will Asterisk, compiled in a 64 bit Linux environment, reap these or other benefits from being on a 64 bit system (other than the system bus speed)? Also, any experience with Asterisk on an Opteron out there? Any unexpected issues? How about card drivers? Thanks! I hope this spurs an interesting exchange of ideas that is of value to many. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Bell HDSL in Ontario good solution for VOIP?
Robert, Honestly, it's better to get colo at 151 Front St. from any big ISP company. Robert Augustyn wrote: Hi, Have you tried it? Any comments would be greatly appreciated. I can have it at C$200, is that a good price? Thanks a lot. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
I met Sangoma guys this Tuesday, and got a AFT102 for evaluation. Right now I am in progress to develop a * channel driver for AFT10* devices. In that case you will have much more flexibility and to use all their API. Steven Critchfield wrote: On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote: David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated I betcha Sangoma has something that'd do this for you. They've been supporting T1 data on Linux for years, and they're recently added zapata to their list of open-source drivers. Give them a shout, they love this kind of stuff. Of course when you go to using the Sangoma cards with asterisk, it appears you lose any extra functionality Sangoma built into the card. That isn't a bad thing, but it negates any benefit of longevity. As for the original posters question. The TE cards from Digium can take care of your ISDN dial ups by itself. Asterisk can't take care of your analog dialups yet. The first thing to know is that you are not splitting the PRI, you are routing calls. Until you get the setup messages, you don't know what is what. Then when you get it, the call could be on any of the B channels. But once you get it, you can determine by the phone number that was dialed how to route the call. You can assign a DID for your dialups and route it all to your portmaster through a separate span or assign different numbers for ISDN and analog dialups so only the modem users go to the portmaster while your ISDN users are handled on the asterisk machine. All others are voice and dealt with from inside asterisk. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Robert, It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. If you are from Toronto ( I suppose you are ), you can check my posts to TLUG (Toronto Linux User Group) regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. I never saw it with Postgres. At the same time Postgres provides you a very flexible SQL language and features, as well as you can make stored procedures on Perl and many-many more. All the Best! Sergey. Robert Augustyn wrote: NICE! I understand that it works against Postgress, any idea what it would take to port it to mysql if anything? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Wednesday, January 26, 2005 12:05 PM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zero Warn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunk note : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belaïd Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCardApplication forAsterisk
Probably. But I believe my eyes and my experience, but not your words. I just respect them, but not trust them. I saw what I saw. You always can compare two of them at the same task loads, and compare performance and stability under heavy load. MySQL licensing policy makes me crazy, their bugs makes me worry, therefore after 3-4 years of using it I decided to stay away of MySQL. MySQL it's like a Sybase/MS SQL in proprietary world. I am trying to stay away from them. I do have a quite good experience with most of the common close/open source DBs, therefore I have some rights to judge them. Brian West wrote: It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. If you are from Toronto ( I suppose you are ), you can check my posts to TLUG (Toronto Linux User Group) regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. I never saw it with Postgres. At the same time Postgres provides you a very flexible SQL language and features, as well as you can make stored procedures on Perl and many-many more. RIIIGHT it sounds like someone doesn't know what they are doing. I have NEVER EVER had anything bad happen to mysql under heavy load. bkw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Vahan, Probably I will go there. But I already ported all my infrastructure from MySQL to PostgreSQL. :) Vahan Yerkanian wrote: Sergey, You should really revisit MySQL.com :) 4.0.x is way outdated... Regarding the high load etc... how about this copy-pasted excerpt from phpmyadmin? ---8 This MySQL server has been running for 19 days, 6 hours, 8 minutes and 28 seconds. It started up on Jan 07, 2005 at 02:21 PM. Server traffic: These tables show the network traffic statistics of this MySQL server since its startup. Traffic ø per hour Received 2,579 MB 5,714 KB Sent 1,050 MB 2,327 KB Total 3,629 MB 8,040 KB Connections ø per hour % Failed attempts 83 0.18 0.00 % Aborted 1,416,484 3,065.05 6.42 % Total 22,064,865 47,744.87 100.00 % Query statistics: Since its startup, 68,530,509 queries have been sent to the server. ---8 regards, Vahan Sergey Kuznetsov wrote: Robert, It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. If you are from Toronto ( I suppose you are ), you can check my posts to TLUG (Toronto Linux User Group) regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. I never saw it with Postgres. At the same time Postgres provides you a very flexible SQL language and features, as well as you can make stored procedures on Perl and many-many more. All the Best! Sergey. Robert Augustyn wrote: NICE! I understand that it works against Postgress, any idea what it would take to port it to mysql if anything? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Wednesday, January 26, 2005 12:05 PM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumberthe Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination numberIt calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zeroWarn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunknote : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belaïd Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Okay. I just shared my experience with MySQL, Sybase, MS-SQL. I never had any issues with Oracle (my application works with DB which stores more than a 1 billion records and still selects the data within 5 seconds quota, at the same time the backend servers inserts the data in that DB every second) and PostgreSQL(not a huge DB, but have very noticeable workload). Steve Prior wrote: Sergey Kuznetsov wrote: Robert, It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. All the Best! Sergey. There might be reasons to prefer Postgress over MySQL, but I find it hard to believe that scalability is one of them - I mean we're talking about the database that runs Slashdot which is so scalable that users reading it routinely take down other websites with the load. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Okay. It was opinion, and I am not demanding to set course to use PostgreSQL. Client of mine had such issue, and even version upgrade didn't solve it. May be they was unlucky with that, but because of that issue they was really close to loose their big customers. Even my experience with MySQL didn't help a lot. The good DB won't screw up with data corruption, even if programmer did it. This is my point of view. Right after that when I started to plan my new project I decided to use PostgreSQL. Not only because of that issue, but because of their weird licensing policy. Just ask why Mark trying to split MySQL module from main branch. Russell Horn wrote: It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. Oh come on, there are many reasons to use Postgres, but this is just FUD. Just as an example off the top of my head, take a look at http://www.livejournal.com/stats.bml (2.5 million active accounts, 367,000 updates in the last 24 hours and all on a mysql backend). There's a host of other big sites all using MySQL - Yahoo! Finance, Slashdot (handling 360 queries per second) and others. If you're losing data on MySQL with 10 users you have a configuration or coding problem. Again, Postgres offers many features that MySQL does not and vice versa, but to suggest that MySQL shouldn't be used because you'll loose data is a bogus argument. Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Athlon 64 for Asterisk?
I am using dual Opteron 2GHz rack-mounted server. It works under Gentoo 64-bit. All applications is native 64-bits. The only issue I had with 64-bit version of G.729 drivers, but I was able to fix it with some LD_PRELOAD script magic. All the Best! Sergey. C F wrote: I'm running an Athlon 64 for * and it works fine. On Mon, 24 Jan 2005 12:16:41 -0600, Carlos Chavez [EMAIL PROTECTED] wrote: I want to buy a new server to run Asterisk and after looking at prices for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64? -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in South Ontario?
MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. Andrew Kohlsmith wrote: First things first -- don't reply to a message about something COMPLETELY different, erase everything and start your new message. Just click on the "To" and start your new message. When you reply and erase everything you are unintentionally placing your message in the middle of an existing message thread. This causes your message to get "buried" and far fewer people actually see it. You don't see this because you are using a mail client that has no concept of message threads. http://www.mixdown.ca/~andrew/dump/threaded_email.png is what a mailing list looks like to most people, and you can see why replying to a message, erasing its contents and starting an entirely new email about a different topic is frowned upon (yours is the highlighted message). Having said that, to your answer: On January 21, 2005 12:20 am, Robert Augustyn wrote: I am looking for a good provider of T1/PRI in Windsor, Ontario. You have many options in large cities. Bell, Group Telecom(360 networks), ATT(Allstream), Telus, Sprint, MCI(UUnet)... There may also be a dozen more "little guys" in your area. Get a few quotes, I find Bell is actually half-assed competitive when they have to be. Things to consider in your quotes received: - inbound or two-way call completion - Number of DIDs per DID/PRI order - # of #s received for incoming calls (4, 7, or 10 usually) - If they restrict the PRI signaling in any way - telephone number "fallback" if the PRI is down (i.e. where do the calls go) - 911/e911 - capability to set callerID/ANI to any DID you are leasing - ability to port existing numbers to the PRI as DIDs - charges for changing anything above once set up -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some issues with X-Lite and codecs.
Yes I did. The same. It looks like there is some packet loss on the way to my VoIP box. Is there any optimal settings for jitter buffer for * ? All the Best! Sergey. Andrew Yager wrote: Hi Sergey, Have you tried phoning from X-Lite to your PSTN line, or your PSTN line to X-Lite? How is the audio quality then? Does it vary depending on the codec you have used? Andrew On 23/01/2005, at 4:31 PM, Sergey Kuznetsov wrote: Hi there, I am experiencing some issue with X-Lite. When I am calling over the phone thru my PSTN-to-VoIP gateway internationally using G.729 the quality is just perfect. When I am using X-Lite to connect the same box, and then to call internationally - I am experiencing some issues. I have 5Mbit/800Kbit cable link with average 60 msecs to my VoIP box. The transfer rate is never falling below 500Kbytes/sec. Therefore I am not suspecting quite noticeable packet loss. I enabled G.711 ulaw, alaw and speex codecs on both sides. By playing with different codecs I am trying to avoid some clicking and sound distortion, which is I am experiencing right now. Speex sometimes is better than G.711, but still having the same glitching. My question is, is there any way to fix it by playing with some parameters on * side, or it's better to play with X-Lite parameters? All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?
Sorry, I completely forgot. You have to have an experience how to use the CRTC site =) If you will click to "Public Proceedings" at the top of the main page you will be redirected to the page witch will show you the most of the useful information. At that page in the "Telecommunications" Part of the table you will see link "Tariff" with is going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm At that pages you have to choose year and then the name of the company you are interesting about. There is the some info buried there, but it's quite easy to find it. I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs from Sprint. http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates. Probably there is some new rates. Have to go thru all recent years. All the Best! Sergey. Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?
I got my PRIs from ISPtel as an add-on to my colo with MCI and thru
MCI. I'll try to find ISPtel web-site (if it's exists) thru
MCI's customer service. Actually Allstream's PRI will cost you around
700-750 CAD per month. It's not that bad.
I got just few PRIs with set of DIDs I need. This is enough for me. I
can set any ANI/C*ID form my range on my PRIs.
My incoming DNIS is 10-digit length.
I didn't try if I can port existing DIDs from another ILECs/CLECs.
All the Best!
Sergey.
Robert Augustyn wrote:
Thanks
You sure have to have experience
...:)
Do you know how I can contact
ISPtel?
Sprint quoted me a realy high
number.
btw: what do you get with your
PRI service?
robert
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Sergey
Kuznetsov
Sent: Sunday, January 23, 2005 5:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
providerin SouthOntario?
Sorry, I completely forgot. You have to have an experience how to use
the CRTC site =)
If you will click to "Public Proceedings" at the top of the main page
you will be redirected to
the page witch will show you the most of the useful information.
At that page in the "Telecommunications" Part of the table you will see
link "Tariff" with is
going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm
At that pages you have to choose year and then the name of the company
you are interesting about.
There is the some info buried there, but it's quite easy to find it.
I cannot find the website of ISPtel either. But I have the PRIs from
them and it's 2 times cheaper then PRIs from
Sprint.
http://www.crtc.gc.ca/8740/frn/2002/a4.htm
- Allstream (ATT) rates.
Probably there is some new rates. Have to go thru all recent years.
All the Best!
Sergey.
Robert Augustyn wrote:
Sergey,
Thanks for the input.
I looked at the crtc site did
few searches but I guess I do not know what to look for because I did
not find anything related to tariffs.
On the same note I am not able
to find a Isptel web site either I guess it is not my day today :)
robert
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Sergey Kuznetsov
Sent: Sunday, January 23, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
provider in SouthOntario?
MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.
The best way to find rates for such lines to go to CRTC site and check
the tariffs for that.
All the Best!
Sergey.
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Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario?
You are very welcome! All the Best! Sergey. Robert Augustyn wrote: Thanks for your help Sergey. robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 8:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario? I got my PRIs from ISPtel as an add-on to my colo with MCI and thru MCI. I'll try to find ISPtel web-site (if it's exists) thru MCI's customer service. Actually Allstream's PRI will cost you around 700-750 CAD per month. It's not that bad. I got just few PRIs with set of DIDs I need. This is enough for me. I can set any ANI/C*ID form my range on my PRIs. My incoming DNIS is 10-digit length. I didn't try if I can port existing DIDs from another ILECs/CLECs. All the Best! Sergey. Robert Augustyn wrote: Thanks You sure have to have experience ...:) Do you know how I can contact ISPtel? Sprint quoted me a realy high number. btw: what do you get with your PRI service? robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario? Sorry, I completely forgot. You have to have an experience how to use the CRTC site =) If you will click to "Public Proceedings" at the top of the main page you will be redirected to the page witch will show you the most of the useful information. At that page in the "Telecommunications" Part of the table you will see link "Tariff" with is going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm At that pages you have to choose year and then the name of the company you are interesting about. There is the some info buried there, but it's quite easy to find it. I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs from Sprint. http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates. Probably there is some new rates. Have to go thru all recent years. All the Best! Sergey. Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some issues with X-Lite and codecs.
Hi there, I am experiencing some issue with X-Lite. When I am calling over the phone thru my PSTN-to-VoIP gateway internationally using G.729 the quality is just perfect. When I am using X-Lite to connect the same box, and then to call internationally - I am experiencing some issues. I have 5Mbit/800Kbit cable link with average 60 msecs to my VoIP box. The transfer rate is never falling below 500Kbytes/sec. Therefore I am not suspecting quite noticeable packet loss. I enabled G.711 ulaw, alaw and speex codecs on both sides. By playing with different codecs I am trying to avoid some clicking and sound distortion, which is I am experiencing right now. Speex sometimes is better than G.711, but still having the same glitching. My question is, is there any way to fix it by playing with some parameters on * side, or it's better to play with X-Lite parameters? All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Becoming a VOIP provider
As far as I understand, if you completely doing VoIP without any PSTN intervention, in this case it's probably unregulated. I case of PSTN-to-VoIP gateway - this is completely different story. Here, in Canada, you have to have International Basic License Class A, to provide (excerpt from Conditions of License Class A, http://www.crtc.gc.ca/INTERNET/1999/8190/Com-Doc/cond-a.htm ): 2. The licensee shall retain until future notice all data with respect to basic international traffic that the licensee (i) transports between Canada and another country using circuit switching protocol on telecommunications facilities operated by the licensee, whether those facilities are owned by the licensee or leased by the licensee from a separate facilities provider, and/or (ii) converts from circuit-switched minutes originating in Canada to non-circuit switched traffic, or converts from non-circuit switched traffic to circuit switched minutes terminating in Canada, regardless of whether the licensee is responsible for the international transport. The licensee is to retain data so that it can provide details of the international traffic, broken down by the number of outbound (Canadian originating) and inbound (Canadian terminating) minutes, indicating the country of ultimate destination or origin. As you can see it covers definition of PSTN-to-VoIP gateway for international traffic. All the Best! Sergey. Steven Wang wrote: I heard about this statement several times. How does it tell whether it is regulated or not? thanks! steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Chad Whitten Sent: Wednesday, January 19, 2005 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Becoming a VOIP provider In the US, VoIP is currently an unregulated information service, not a regulated communications service so things like CALEA and E911 can just be overlooked if you choose. On Wednesday 19 January 2005 14:19, Ed Robbins wrote: Manjit Riat wrote: That was a really nice description... Can you do 1-14 and I'll do 15 and 16?? Just kiddin. -Original Message- From: Ty Carter [mailto:[EMAIL PROTECTED]] Sent: Wednesday, January 19, 2005 10:58 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Becoming a VOIP provider 1. You must have some type of business model / plan 2. Be well capitalized, starting out is going to be a cash draining experience. 3. Have access to (U.S.) PRI or Channelized T1 and High speed Internet connection 4. For U.S. it always helps on the bottom line if you're a CLEC 5. Have a test server, if you want to play in the enterprise market, buy a test 1U server and a 1 T1 PRI card 6. Forumlate your POPS 7. Get a ANCP Code from Telcordia, then apply for a CIC, Part A code (commly reffered to as a PIC code (10-10-987) 8. Arrange for a LD carrier, preferabably one that can terminate and originate via SIP, IAX or IP 9. Arrange for PSAP integration/handoff (for 911) 10. Have your lawyer establish your Terms of Service and disclose to your clients about the 911 availability and have them sign off on this. 11. When all of the above is satisified and working, formulate a beta pool of clients, a couple of small businesses and a few residentials 12. Give them cutrate service for testing 13. Once your have your beta trials, put it into production and let the money start flowing. 14. Put in a HP Blade server rack, and start provisioning asterisk like crazy. 15. Laugh all the way to the bank 16. Retire when your 47 and relax on the beach with a beautiful woman in one hand and a cold drink in the other :-) That is about all there is to it. Any more questions? Ty Carter Strategic Network Consultants, Inc. 524 East 9th Street Washington, NC 27889 [EMAIL PROTECTED] P.S. The last few items are just a joke.. Please, list, don't bombard me with flames about hardware vendors or laughing on the way to the bank. This is just a 30,000 ft overview. If you want specifics, contact me off list and I will try and help you. I don't know applicability in Australia, but in the US don't forget about CALEA. Seems like that is a big issue for a lot of providers to come to terms with. Ed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chad Whitten Network Administrator neXband Communications [EMAIL PROTECTED] 601-944-4801 Phone 601-944-4803 Fax ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] X-Ten lite troubles.
Hi guys, I do have some weird situation. I do have an * box, and I want to connect to that box from my Windows box by SIP via X-Ten Lite. I made configuration of that soft phone as it was suggested by lots of tutorials I found by Google. But... it doesn't work! I don't know what is wrong there, but I have unobstructed access to my asterisk box, created user in sip.conf, enabled 'sip debug ip' but there is no any response at all. When I dial number soft phone saying 'Call not approved'. How can I get rid of it? Can someone provide me an example for X-Ten lite (user + password) specifically for Asterisk I will be very appreciated. PS: At the same time my SIP hard phone works well with *. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?
I would be interested in this list as well. I have an positive experience how to get License Class A from CRTC. As well as I am interested to talk about LNP portability. All the Best! Sergey. Andrew Kohlsmith wrote: On January 17, 2005 04:47 pm, Jim Van Meggelen wrote: LOL. I hadn't thought of it that way. Little vignettes amidst the commercials? Exactly -- It's precisely why I hang around on linux-elitists and a couple other oddball lists... a good 90% of what's there is crap but man when something good comes by... wowza. Just because the volume isn't there? That might be a good thing, ya know - have a list with, say, one or two messages a day, on average. True, but that's why I like looking at sineapps now and again -- they sometimes focus on things that I've not even seen, it's interesting reading... but I slug it out on the list well, just to slug it out. :-) It was a policy at our company that any new product implementation would always require Technical Support be involved until several engineers, technicians and installers were comfortable with it. I hope I always remember the lessons learned from getting new products, and having to develop training and implementation practices. ... Seems that making mistakes is actually a fantastic (albeit uncomfortable) way to learn. I sometimes wonder if I unconsciously muck things up at first as a rite of passage. We have a similar policy here and it really helps people understand why things are done a certain way when they have to field some of the customer calls themselves. "right" and "wrong" take on new nuances that they would have otherwise been oblivious and even belligerent towards. Nobody knows a thing so well as those who can expertly break it. That sounds very close to "As soon as you make something idiot-proof along comes a better class of idiot." :-) Would it be considered trolling to start a thread on Cleaning Maple Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? Let's not forget the weekly "tooques and telephony" segment, and a review of the best block heaters for your wi-fi fones. Does that mean I'm right and you're wrong? Yes... oh, wait... Aughhh! -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canadian Content: Telus and Shaw...
If they will do it, you are welcome to write the letter to CRTC and other governmental agencies for uncompetitive behavior. I think it should work. All the Best! Sergey. Kim Lux wrote: I called Telus before Christmas requesting some sort of VOIP connection. Here is what I learned: a) the guy I was talking to never heard of * b) they didn't think there was any way that a PC could perform the duties of a PBX c) they told me they didn't have any VOIP connections, but then told me that they would supply and connect Nortel PBXs using H323 d) they would not supply me with an H323 connection for *. I don't have time to discuss this in detail, I just thought I'd share it based on the chat in the CDN list discussion. We are going with babytel. I'll advise how that works when it is up and running, hopefully next week. BTW: Shaw is supposed to start supplying VOIP on a separate network from their high speed network. Here is the news clip: http://www.canoe.ca/NewsStand/CalgarySun/News/2005/01/14/898082-sun.html I find this interesting because several people have told me they are using Shaw's high speed Internet service as the backbone of their VOIP system. (Extreme is supposed to work even better.) I wonder if Telus is going to block the SIP ports on their ADSL network ? I wonder if Shaw will ? (Telus presently blocks the SMPT port so that you MUST you their mail server.) I wonder if shaw or telus people lurk on this site. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?
In my AGI script I made the next trick:
$digit = $AGI-get_data(vm-enter-num-to-call-then-pound, 15000, 1);
while ( $digit eq 0 or $digit )
{
$phoneNum .= $digit;
$digit = $AGI-get_data(empty, 7000, 1);
}
where file empty.gsm have 0 byte length.
It works like a charm for me.
All the Best!
Sergey.
Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
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