RE: [Asterisk-Users] g729 quality at GSM bitrates
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, February 20, 2006 11:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] g729 quality at GSM bitrates I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. To my ear, ILBC sounds much better than GSM. It's slightly more efficient, and more tolerant of things like packet loss. Some folks, hate the sound of ILBC encoded calls. shrug Your other choice would be G.726/32. * supports it, as do many ATA's and softphones. It's a bit fatter, but sounds MUCH better than GSM. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Corlett Sent: Tuesday, February 14, 2006 9:01 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Solution for 1 time blast of 200,000 recorded calls Ron Senykoff [EMAIL PROTECTED] wrote: I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Can you tell me which party this is for, so I can ensure I never vote for them? Do your fellow citizens a favor and just don't vote, period. Anyone who would make a decsision as important as voting for a political office or issue, based solely on party affiliation or worse, the questionable antics of one individual (with plans to VOIP-SPAM 200,000 people), can not be trusted with such an important responsibility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do we need a QOS switch ?
stoffell wrote: On 2/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. If you don't overload your internal network, you'll be fine.. Ah... THERE is the key phrase we were looking for. The proposed VOIP traffic will have little impact on the usability of their network FOR VOIP traffic. It is all the other stuff that runs across their LAN that make make VOIP a really cappy idea, if the don't take steps to ensure that the VOIP traffic is managed properly. With the paucity of details provide by the OP, it is impossible to say, with any degree of credibility, that the ...will be fine... Do those 10 phone sit on the desks of graphic designers, whose file and print traffic can bring a 100 Mbps segment to its knees? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk video conference
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shain LeeSent: Thursday, February 02, 2006 2:14 AMTo: AsteriskSubject: [Asterisk-Users] Asterisk video conference Hi , I just wanted to know , how would be asterisk work with video calls ? What are the hardware do we have to buy ? Who are the providers of particular harwares ? Can we use video calls / video conferenceing in the LAN perfectly ? How it would be depends on the WAN ? Asterisk's support forvideo over SIP is very rudimentary. Only to video codecs H.261, H.263, and H.263+ are supported, andeven then, not very well. There is no support for dynamic negotiation of frame rates, etc. Queries to the -dev list, as to progress on these features were recently met with silence. We will be looking to jump into the project to support our own initiatives in the area of video in a few weeks. Until things change, your best bet for connecting SIP video phones is SER. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, January 04, 2006 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. What kind of loads are we talking about here, please? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Server Specification
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Code Lover Sent: Thursday, January 12, 2006 1:39 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Server Specification Hello, Is the hardware specification is enough to get 300 simultaneous calls? What should be the Bandwidth to get 300 simultaneous calls? What do you mean by get 300 simultaneous calls? If you plan on using that platform as a PSTN gateway, using multiple Digium TE4xxP cards, the answer is probably not. Assuming that you will want to save on bandwidth costs by using a codec that provides a substantial degree of compression (G.729), there's no way that the spec'd box will transcode 300 channels. If Digium's (rather sparse) dimensioning information is to be believed, you'd need three such boxes to terminate that load. On the other hand, if your box will simply be serving as a gateway to gain the significant bandwidth savings afforded by IAX2 trunking, and relying on other hardware to actually terminate the calls, then yes, it could do handle that load and more, since it is functioning as little more than a router in that role. IF you are able to use asterisk and take advantage of IAX2 trunking, and using G.729, you could do 300 simultaneous calls with about 3 mbps. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Non-PRI T1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, January 06, 2006 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Non-PRI T1 Are they configured for inbound calls? If so how? Usually the telco sends the last 4 digits of the called phone number down the line. Uhm, don't you need PRI signalling for that? -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.14/222 - Release Date: 01/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] prepaid application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of scott
Sent: Wednesday, November 30, 2005 11:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] prepaid application
Hi All
I am using prepaid auth (callingcards), the idea is for a
prepaid support line. It is up and running but I have a
couple of questions with regards to modifications I would
like to make.
When a user calls and they go through the process of entering
their card number. They are then asked for a destination.
What I would like to be able to do is not have it ask for a
destination and automatically dial a number?
How about something like:
exten = 1234567,1,read(CARDNUM,promptfile)
exten = 1234567,2,agi(astcc.agi,${CARDNUM},5566)
...where promptfile is the name of the prompt instructing the caller
to enter his account number, ...followed by the pound sign, and 5566
is the extension you want dialed after authentication.
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.10/189 - Release Date:
11/30/2005
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RE: [Asterisk-Users] 911 Notices
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Friday, August 26, 2005 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Notices After that date passes, technically, you're supposed to offer it if you're business is interconnecting voip networks to the PSTN. ___ An important distinction should be clarified here. The FCC will require interconnected VOIP providers to provide 911/E911 service. Those VOIP operators providing dial-tone and a DID number need to comply. Those providing termination, or origination alone will not. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/82 - Release Date: 08/25/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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
RE: [Asterisk-Users] Limiting the number of calls
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Miron Sent: Wednesday, August 10, 2005 11:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Limiting the number of calls Hey everyone. I'm wondering if anyone has any ideas on a way to limit the number of outbound calls at a time, and if the limit is reached a message is played when someone tries to place the next call. I've searched the wiki but have yet to come up with anything. Search again, for SetGroup and CheckGroup commands. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.5/67 - Release Date: 08/09/2005 ___ 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] Astcc Charging \ Matching Pattern Problem
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ade AgberoSent: Tuesday, August 02, 2005 2:32 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Astcc Charging \ Matching Pattern Problem Astcc applies a charge for Czech Republic - Mobile Code- 4207 to a call destined for UK Landline 44207. It appears Astcc uses the first matching pattern of4207 it finds in the routes table instead of continuing to search through the routes table until it comes to 44207 for UK. Any ideas on how to resolve this problem. Remember, ASTCC is evaluationg the number string as aregular _expression_.Without the ^ character prepended to the string, you'll get a match on thatroute no matter wherethatroutesnumber stringmight exist in the dial string. ^4207 means "match only those strings that START with 4207", whereas 4207 means "Match any string that has 4207 anywhere in the string." -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 08/01/2005 ___ 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] Question about Nextone softswitch
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 27, 2005 1:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Question about Nextone softswitch As an exampleif we have a call that: originates via PSTN line to one of our local DID's in Seattle comes into our Asterisk server in Los Angeles or Denver is routed by Asterisk for termination back to a different Seattle PSTN and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone softswitch in Dallas before ultimately terminating at the desired Seattle PSTN line... What is the resulting affect as it relates to any difference in user experience for the caller in Seattleand what, if any, is the cost difference on our end due to the extra hop? The extra trip around the country will add significantly to the latency of the voice traffic, probably in the neighborhood of 120 milliseconds, in your example. This is enough to cause problems for some callers. Others (most, probably) won't even notice it. This assumes, of course, that the call doesn't bounce around the IP network some more, after you've sent it to your termination provider in Dallas. As for the cost, assuming that you are paying to have both the origination and termination legs transit your switch via VOIP anyway, I don't see where any additional cost would be incurred by routing it to Dallas. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.6/59 - Release Date: 07/27/2005 ___ 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] To anyone seeking 911 Service Providers stayaway!!!
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Tuesday, July 26, 2005 6:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] To anyone seeking 911 Service Providers stayaway!!! There where people saying this Vonage issue was all FUD, anyway, side effect was...Seems this is 911 for VOIP is FCC mandatory now in USA ? Not sure, I use * at my home and have DSL, so I just route my 911 to the landline outbound, I would not expect the outbound IAX providers to offer 911 to me :-) The FCC regulation is still in the comment phase. When it goes into effect it will require interconnected VOIP lines (defined as those enabling calls both to and from the PSTN) to provide access to the appropriate local PSAP, via the appropriate selective router (where on exists for that location). The regulation further requires providers of interconnected VOIP service to provide one or more methods for end-users to update their location. At least one of those methods must be accessible via nothing more than the CPE. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 07/25/2005 ___ 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] ASTCC: different incriments
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Tuesday, July 26, 2005 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC: different incriments How can I fulfill that? *Billing Increments* Continental USA: six (6) second increments. International: thirty (30) seconds minimum and six (6) seconds thereafter. Mexico: sixty (60) seconds minimum and six (6) seconds thereafter. The billing increment is set in the brands table. When you create cards, this value is copied into the inc column in the cards table. (I'll spare us the rant on normalization here...) The per call minimum is set in the includedseconds column, in the routes table. This value, along with the value of the connectcost column for a given record (route) is used to compute the cost of the call. So, in theory, you set all your cards for 6 second increments, and you set your routes to 6, 30, or 60 includedseconds. That's the theory, but the stock ASTCC code has a bug in the way it makes this computation. Darren has reopened the bug report. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 07/25/2005 ___ 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] To anyone seeking 911 Service Providersstayaway!!!
The mandatory part that is due right now is the section of the law that deals with informing the voip user of their current E911 status. That part is not in a comment phase. Actually, that part takes effect July 29. The access requirements will probably hit at the end of November. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 07/25/2005 ___ 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] ASTCC gives me only the time, but no cost
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wiebe
Sent: Saturday, July 23, 2005 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ASTCC gives me only the time,
but no cost
Thank you very, very much Rusty. I reopened the bug report.
http://bugs.digium.com/view.php?id=4479 I made a very slight
change to
the method it uses to calculate costs but it should implement the
connect charge properly. Initially I rewrote the cost
calculation code
but that was not necessary, it can be implemented by changing the
following lines
my $adjtime = int(($answeredtime + $increment - 1) / $increment) *
$increment
becomes
$adjtime = int((($answeredtime - $numdata-{includedseconds}) +
$increment - 1) / $increment) * $increment
This can yield a negative number, where $answeredtime
includedseconds, can it not?
--
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Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005
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RE: [Asterisk-Users] super high bandwidth codec
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Sunday, July 24, 2005 9:11 PMTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] super high bandwidth codec It has nothing to do with bandwidth. It has everything to do with your routing gear! This is completely incorrect. Skype uses a codec that uses far more bandwidth than traditional telephony provides, which is why it's audio can have morerange than even the best quality phone call. In theory, there is nothing preventing an all VOIP network from using such a codec, but as a practical matter, at least part of most phone calls are via traditional phone gear and/or networks, you don't see it widely deployed. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005 ___ 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] ASTCC gives me only the time, but no cost
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Saturday, July 23, 2005 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ASTCC gives me only the time, but no cost The included seconds field is not taken into account when billing the connect charge. IMHO this is a bug but I've not gotten enough feedback to put the patch through. Therefore the patch has been closed. :-) I spent an afternoon going through that code again, Darren. You were right. If we assume that the intent was to use the includedseconds column value as a way to allow for x/y billing intervals, and set connectcost to the value that we want to charge for the call minimum charge - x, the stock code charges that amount, but also starts the meter running on the y value from the start of the call, resulting in an over charge. The y value, by the way, is set in the brands table and flows to the cards table when cards are created (breaking normalization). For example, we have a route to McMurdo Station for which we charge $.50 per minute, in six second increments with a 30 second minimum (30/6). If we set the connect charge column to $.25, the included seconds to 30, and the cost to $.50, a 30 second call should cost $.25. Instead, it's costing $.50, because ASTCC charged the connect fee, plus the cost of 5 six second increments - $.25. It shouldn't start charging those six second increments until AFTER the includedseconds interval has passed. I've patched by scripts to correct this. It would be nice if the correction were made to the distributed source, perhaps with some documentation of how things are supposed to work. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005 ___ 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] changing Nobody picked up in 30000 m
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of wassim darwish Sent: Friday, July 08, 2005 1:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] changing Nobody picked up in 3 m i dont know how to edit the the time for ringing 3ms to 4ms,please help me. Start here: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 07/06/2005 ___ 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] URGENT: hardware spesifications needed
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael L Smith Sent: Thursday, July 07, 2005 12:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] URGENT: hardware spesifications needed Who are you to decide what Information can and cannot be legitimately be sought here:? Just curious. And opinionated. Which is fine. We are each entitled to our opinions. MY opinion is that lazy jagoffs, who won't lift a finger to learn how to do something and want it spoon fed to them, should not be surprised when their demands are met with rude replies. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 07/06/2005 ___ 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] URGENT: hardware spesifications needed
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Frisch Sent: Wednesday, July 06, 2005 4:22 PM To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed Come on now children. Is this not a place to share knowledge? Well..., yes, and no. Information that isn't readily available elsewhere may legitimately be sought here. However, when the question is of the FAQ variety, and it is clear that the person asking it has not even attempted to find the information for himself, then rude replies are not out of line, IMO. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 07/06/2005 ___ 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] Simpletelecom dead?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: Tuesday, July 05, 2005 11:27 AM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simpletelecom dead? I've gotten word from their Marketing VP. They are doing some kind of massive move and expect to be down until Thursday Dad needed the driveway for the motorhome after the holiday weekend, so they had to move the lemonade stand... -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.9/39 - Release Date: 07/04/2005 ___ 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] VOIP Providers Problems
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Smith Sent: Monday, July 04, 2005 2:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VOIP Providers Problems you guys are so friggin funny.. all i see bout problems on most providers here are users who never read a line of the handbook i could prolly solve all these eyes closed with the asterisk handbook on my side as a friend. wake up.. i work for a hosting provider and we get lots of users assuming they have it all right and us is the problem. and in fact 99% of the time they are the ones who fucked theyre servers up and such. remainder of juvenile rant snipped Wow. Where were you when LiveVOIP needed some good customer service people? You'd have fit right in with that outfit. LOL -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.8/37 - Release Date: 07/01/2005 ___ 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] LiveVoip is Bankrupt
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Sunday, June 26, 2005 11:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt Hi All, I think by now everybody knows that LiveVoip went down, bankrupt etc So please stop nagging about it and move on to some topics that really matter. If you want to discuss LiveVoip, get all together in a restaurant, eat, drink and nag and wine about it as much as you want. But do it there and not here. Thank you, Mr. Self-Appointed Netcop. Now please study the features of your mail client that allow you to avoid reading offensive topics. Granted, this issue is only tangentially topical for the -users list, but I believe the discussion is largely worthwhile, if only for the lessons this episode brings to us. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.2/29 - Release Date: 06/27/2005 ___ 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] HooDaHek 0.2 Released
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Pralle Sent: Monday, June 27, 2005 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HooDaHek 0.2 Released HooDaHek 0.2 Released Just a few changes: - Added MSN Messenger support to the notification bot - Added debugging code to the dbhandler script. http://www.nathanpralle.com/software/hoodahek.html Nathan, What you're doing here is way cool. Thanks. May I suggest adding Jabber support soon? Thanks again. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.2/29 - Release Date: 06/27/2005 ___ 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] 12 FXO ports into Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wright Sent: Thursday, June 23, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 12 FXO ports into Asterisk I have a client that has 10 POTS lines incoming. There is no other option for lines here. I have 3 options I can see: 1. 3 TDM400 cards 2. An external SIP/FXO gateway 3. A T1 card plus a channel bank. Does anyone have any thoughts on these 3 or suggestions on keeping the cost down? For what you would spend on options one or two above, you could by the T1 card and an used channel bank (configured with the requisite number of FXO ports), and have money left over. Ebay is your friend. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.11/26 - Release Date: 06/22/2005 ___ 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] ASTCC Rate Calculation
eval { $cost = int($adjcost * $adjtime / 60) };
#cost = 253
Corrected, this would be 250.
Viewed another way, using a 6 second increment, 147 seconds
represents
25 such increments (actually 24.5, but we get all of the last
increment, so it's 25).
25 * 10 (the cost of one 6-second increment) = 250.
Yes, but we need to allow for 30,6 6,1 60,30 billing. I
think the
easiest/best way to handle this is the connect charges as ASTCC
presently supports them.
I agree that ASTCC is, at present, wholly deficient in managing y/x
billing schemes (anywhere y != x).
I'd rather NOT use the connect fee to do this. If we're going to fix it,
let's fix it right. I don't have time to hammer it out right now, but it
seems to me that as long as y is evenly divisible by x (resulting in an
integer value), it should be pretty simple to come up with an algorithm
that will properly handle things like 30/6, 60/30, 6/1, etc.
--
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Re: [Asterisk-Users] ASTCC Rate Calculation
On Fri, June 17, 2005 5:19 pm, Darren Wiebe said: Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480 I may well be wrong but I believe the issue needs visiting. Somebody was asking me how it calculates costs as they thought they knew what a call should cost. I said I'll show you. Mistake, I could not come up with an answer that made sense. Darren, I took a quick look at the patch. I'm not certain, but it appears that you've taken out the formula that factors in the billing increment. This forumla, inything other than a 1 second incement, will always add time to the call for any number of seconds not equally divisible by the billing increment integer, resulting in a slightly higher cost than might be expected at first glance. This is the way it is supposed to work. As I said, I only glanced at it briefly. Could you describe your changes and the error you were seeing? ___ 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] ASTCC Rate Calculation
On Sat, June 18, 2005 2:34 pm, Darren Wiebe said:
Okay, I'll post both pieces of code. What I was seeing is that calls
where being billed more than I thought they should be. Lets use an
example with the following info:
Call Length: 147 Seconds
Increments: 6 Seconds
Connect Charge: 100
Included Seconds: 30
Cost per minute: 100
1. Present Code:
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment)
* $increment };
#adjtime = 152
This might be where your error is creeping in. $adjtime SHOULD equal 150.
Remember, the int() function removes the value to the right of the decimal
point - so int(($answerdtime + $increment -1) / $increment) = 25 and not
25.3~, as your example appears to show. This makes $adjtime
actually 150, not 152.
eval { $cost = int($adjcost * $adjtime / 60) };
#cost = 253
Corrected, this would be 250.
Viewed another way, using a 6 second increment, 147 seconds represents 25
such increments (actually 24.5, but we get all of the last increment, so
it's 25).
25 * 10 (the cost of one 6-second increment) = 250.
$cost += $adjconn;
#Total Cost = 353
2. My Proposed Code:
$total_seconds = ($answeredtime - $numdata-{includedseconds})/$increment;
#Total_Seconds(This variable is not very well named) = 19.5
$bill_increments = ceil($total_seconds);
#We need to bill for 20 6 second increments.
$billseconds = $bill_increments * $increment;
#This translates to 120 seconds.
Which cheats us out of 27 seconds worth of revenue (actually 30 seconds,
since that 27 seconds represents five 6-second increments).
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RE: [Asterisk-Users] Portable USB headset for VoIP
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Tuesday, June 14, 2005 4:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Portable USB headset for VoIP I've bought bunches of these: http://www.tigernetcom.com/products_USB_100.html they work great. Very handy. That's a HANDset. The OP was looking for a HEADset. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.3/15 - Release Date: 06/14/2005 ___ 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] howto write CDRs on two mysql servers
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Musone Sent: Thursday, June 09, 2005 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] howto write CDRs on two mysql servers why not just use mysql replication to the second one? On 6/9/05, Rosario Pingaro [EMAIL PROTECTED] wrote: For redundancy I would like to write the CDRs on tow mysql servers. I thought of that, briefly, but redundancy would rather dictate that if MySQL Server 1 went down, records could still be created on MySQL Server 2. Not possible in the replication scenario. This would be a nice feature to have, either a second live database connection, or the option of configuring a failover server, to be used in the event that * can't connect to the primary. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 06/08/2005 ___ 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] Voicemail and MS Exchange Synchronization
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: Thursday, June 09, 2005 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this being developed for Asterisk and yet it would appear to be a critical component needed to migrate customers used to fully integrated Unified Messaging systems to Asterisk. (a) Has anyone cracked this nut (or started on it)? (b) Anyone interested if we post a bounty? Good luck! Back in the day, when we were on an Altigen system, we were using this feature. It NEVER worked right. To be fair, it was not a feature that had been extensively tested. Altigen's beta program appeared to made up of paying customers. :( From what I recall of the sessions with their engineers trying to debug things, Exchange Server's behavior in the areas critical to supporting this feature were poorly documented and seemed to change from one service pack to the next. Things may well have improved, with regards to Exchange Server. It's been a few years. To be sure, this would be a killer feature in marketing * to MS Exchange Server shops. But I think I'll go hit myself with hammer for a while instead. :) -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 06/08/2005 ___ 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] Multiple E1s on one box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Wednesday, June 08, 2005 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Multiple E1s on one box Answering both questions: 1) I am connecting to a Meridian usin a SIP E1 Gateway in R2. I just bought the card and one of my test will be direct R2 connection. Have not tried yet. 2) I was told you can do 12 E1 as long as it is G.711, but nobody is telling me how many E1s per box doing G.729. I have read twice that 80-90 ports is possible, but others tell me that no more than 30 is possible. Of course, the biggest one box CPU in consideration is a Dual XEON 3.0 with 1 GB RAM. I believe that 12 E1 is a bit optimistic. Maybe not. I'm extrapolating from the translation times I have at hand here. I do believe Digium's statements that 80-90 simultaneous G.729 -- ZAP conversations is the practical limit on a dual Xeon box. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.5 - Release Date: 06/07/2005 ___ 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] G729 vs. gsm
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Friday, May 27, 2005 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] G729 vs. gsm Well, it does to anyone without hearing damage. It sounds very obviously different. Different, yes, but to what degree is an entirely subjective judgement. Ergo, your judgement of ...very obviously different... is valid only for you. Please do not get me wrong that G711u sounds better through the PSTN. Thats a given! You can't convert G729 up and down to G711 and expect the sound quality to be there. This is meaningless drivel. Hardly. Each conversion introduces the equivalent of gen loss. Two such conversions are easily encountered, especially when dealing with a third-party network, and will produce (in MY subjective opinion) positively crappy sound. Since it doesn't correlate with the impression of even the developers of G.729, it *is* bad information. Realistic people know G.729 will be worse. What they need is meaningful guidance as to just how much. Yes, and your guidance is oh-so-meaningful. roll eyes -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Friday, May 27, 2005 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) of client hardware. Could you use javascript, or java from within the browser, which is both portable, and likely to work on ANY browser that way there is no installation as such, just visit the page, and leave a browser window open (minimised) which is 'listening' for connections ?? Sigh... Browsers don't listen. They inititiate a connection, process the requested transaction with the web server, and close the connection. The simply can't be used to listen for an arbitrary connection. I have to agree your way takes up less resources, but if you modify my agi script to write XML file instead of putting the data back in a DB, the load will be close to 0 (never seen a current webserver that cannot do less then 1000 xml file serves per second). Again, this is not the way to do this. Dozens, or hundreds of clients constantly hammering a server with Have you got anything new? No? OK... messages every couple of seconds is an excellent example of how NOT to design a system. Yes, you can get away with it, if the resources involved are not an issue, but I think it fair to assume that for many interested in this discussion, resources like bandwidth and CPU usage ARE issues. At the end of the day, I'm sure we all agree that a push method is best. Quite. Alas, that means an instance of SOMETHING on the client that can listen for, and respond to, arbitrary events. Personally, I don't know enough about all these scripting languages etc, but if it is possible, then that would be wonderful :) About the closest we are likely to come is with a Java applet. Even then, there are a lot of environments that won't allow that solution either. -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Fanning Sent: Saturday, May 28, 2005 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) Could you use javascript, or java from within the browser, which is both portable, and likely to work on ANY browser that way there is no installation as such, just visit the page, and leave a browser window open (minimised) which is 'listening' for connections ?? Sigh... Browsers don't listen. They inititiate a connection, process the requested transaction with the web server, and close the connection. The simply can't be used to listen for an arbitrary connection. Actually, I don't think that you are quite right here. The guy mentioned Java from within the browser. I believe that I am right in saying that a Java applet should very well be able to listen for tcp connections as well as udp D'oh! I had misread the PP's statement and assumed he meant a bareback browser window. You are, of course, quite right. A Java app could handle this, but we are still left with the issue of having to install SOMETHING, even if it is a small Java app, on the client to make this work. -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Fanning Sent: Saturday, May 28, 2005 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) No installation as such, just make sure a Java virtual machine is present on the machine. Seconds to load. Okay, we're splitting hairs here about installed versus simply DL's and launched by the JVM, but the point is that Java must exist and be enabled on the client to support this. I agree that this is a reasonable assumption, but in a few minutes, someone is going to post an objection that you can't make that assumption... and she will be (technically) correct. Still... I would say that Java would be ideal for an application like this. I agree completely. It seems like the best platform choice, by a substantial margin. -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] VoiPSupply Dot Com
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory AndrewsSent: Thursday, May 26, 2005 6:33 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] VoiPSupply Dot Com Karl, first off I apologize for any inconvenience on your recent order. I will take a look at your transaction to see where things may have gone awry. We do make mistakes, but we strive to not make the same mistake more than once. Secondly, I apologize to the list moderator for the pseudo-commercial nature of this post. The grievance was aired on this list, and I felt compelled to respond to this list and I realize much of this may be more appropriate for the BIZ list. Cory, You sir, are a class act. The message quoted above (snipped, for brevity) is an excellent example of how customer relations should be handled. While it appears that thethe issue was, at leastin part, dueto some less than effective business processes on your end, as well as a partially clueless customer,you handled the customer with courtesy and respect; the hallmark of a company thattruly VALUES their customers. Clearly, you understand that it is the interest of the business to make those customers happy. Some of the other vendors on these lists would do well to pay attention to the lesson that Cory just gave. -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] Newbie here. Tips on setting up 100 phones wanted.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brandt Milczewski Sent: Friday, May 27, 2005 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted. I'm looking at setting up Asterisk for a completely IP environment. All intercompany calls. I work for a ski area. I currently use a 3Com Superstack for in our office. And an old small town phone system for up at the mountain. The phone system is dying and I'm hoping to bring IP to replace the old phones. It will be about 100 phones at about 20 locations all within about 4 miles of each other. With a run of 4 miles, its a safe bet that some segments will be (as others have pointed out) beyond the 100 meter limit of ethernet. In practice, under optimal conditions, you can fudge this a bit (sometimes a lot) but I wouldn't count on it. For the longer hauls, you might want to consider point-to-point DSL, using something like this http://www.paradyne.com/products/SNE2000/ Conceivably, if you own the copper, you could do anything you want with it. -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, May 25, 2005 7:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) It doesnt seem to be complicated but for example, the things that bother me are refreshes, I dont want to use meta refreshes for this monitoring webpage every X seconds, rather, use something more realtime... Any ideas? And that's the real trick. Web browsers, unless they are instructed to do otherwise, don't DO anything once they've completed loading a page. So without instructing them to refresh, they aren't going to be aware of a server-side change, such as an incoming call. For that, you're going to have to have some way of sending a message TO the client machine, have it received by that machine, and have that client machine take the desired action (pop up an incoming call dialog, load a contact record, etc.). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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 FXS modules too fragile?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Wednesday, May 25, 2005 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium FXS modules too fragile? SOME people also puzzle over the fact that you can't boil eggs on an electric guitar. Of course you can. Ever heard of Jimi Hendrix? Heh-heh... right. I think I'm calling up the same image that you did, though technically, at that point, it became a naptha guitar. At any rate, the guy was certainly versatile, wasnt he? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan PerikSent: Wednesday, May 25, 2005 12:13 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] CRM integration (was RE: CallerID)Rusty Shackleford wrote: And that's the real trick. Web browsers, unless they are instructed to do otherwise, don't DO anything once they've completed loading a page. So without instructing them to refresh, they aren't going to be aware of a server-side change, such as an incoming call. For that, you're going to have to have some way of sending a message TO the client machine, have it received by that machine, and have that client machine take the desired action (pop up an incoming call dialog, load a contact record, etc.). http://wp.netscape.com/assist/net_sites/pushpull.htmlWould that work? Especially the "server push". Not sure if current browsers like it or not. I've never tried it, but came across this document, and thought it may be something useful. Apparently not. At least not with Firefox, as the demo doesn't work. Also, though I didn't spend a great deal of time analyzing the stuff there, it appears to have the potential to also generate an unacceptable load on the web server's resources as the number concurrent connections increases. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] CRM integration (was RE: CallerID)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Wednesday, May 25, 2005 12:04 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) And that's the real trick. Web browsers, unless they are instructed to do otherwise, don't DO anything once they've completed loading a page. So without instructing them to refresh, they aren't going to be aware of a server-side change, such as an incoming call. For that, you're going This is not true. I beg to differ... Please re-read my statement that ...unless instructed to do otherwise... If it was for pure HTML only, yes, you are correct. But with javascript you can start a timer and execute a javascript function every once in a while. If this javascript loads an XML document off the server, you're there ;) So you have now instructed the browser, via javascript, to periodically poll the server every once in a while. This is exacly what the previous poster (the one I replied to) was trying to AVOID, and for good reason. It doesn't scale. In order to be effective as a way to present the user with caller-ID driven data, it would have to poll quite frequently. With a handful of clients constantly doing this, the impact is inconsequential, but as the number of clients hammering the server in this manner climbs, things are going to break. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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 FXS modules too fragile?
Yes, one might think that, IF one didn't understand the nature of electricity and electrical components. SOME people also puzzle over the fact that you can't boil eggs on an electric guitar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ian Pattison Sent: Tuesday, May 24, 2005 4:00 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Digium FXS modules too fragile? One would think if it can generate it it can survive it as well [EMAIL PROTECTED] 24/05/2005 04:07 Ian Pattison wrote: Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage. AHAHAHA You burnt your FXS port! No they can't survive 89 volts! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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] LOOKING TO HIRE
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Thursday, May 19, 2005 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] LOOKING TO HIRE Or rather, let me take that back. If you do not recognize the value of Perl or Python as appropriate, valid programming tools for certain scenarios (for example, prototyping AGI scripting with Perl), I doubt that /you/ are what I would consider a good programmer. I'm building something around an industrial SBC with the built-in tiny basic intepreter. I guess I'm not a good programmer, huh? Why..., you're no programmer AT ALL! mutter ...stinking hardware-hacking cretins... /mutter -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ 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 NCFA numbers
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Dziubanski Sent: Wednesday, May 11, 2005 7:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] broadvoice NCFA numbers Im about to drop their service; looking for another service that allows asterisk and has in bound AZ 480 dids right now. It should say And our ENGINEER not ENGINEERS are hiding under their desk hoping the problem will go away, we highly doubt to have this resolved within a week, please don't call, we don't answer out phones In all fairness, they do answer their phones, and (in rather stark contrast to some other VOIP providers) their support staff are remarkably pleasant under what must be extremely trying circumstances. The wait is long, to be sure, but there are humans there. The human I spoke to took time to do what trouble-shooting he could, was grateful to have a clue-ful asterisk user that could at least tell him what errors were being returned, and then eventually explained that the problem was with a vendor and that the ETR supplied by the vendor was long passed. He could offer no realistic estimate for restoration of service. It's a shame that they can't get things fixed. This episode is going to cost them dearly, because I believe that lots of others are doing what I'm doing, and pulling the plug. It's been a week now, and there are other options. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 05/10/2005 ___ 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] Voicemailbox on Queue?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Sent: Wednesday, May 04, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemailbox on Queue? Thanks, Paul for the info. What I'm try to find out is if there is a voice mailbox for the queue, so that all members will be notified of waiting messages, and any member can check and manage the voice mail box. Within the context you specifiy in queues.conf, you must provide an extension to handle the digit that the caller is directed to press in order to leave a voicemail. For example, let us assume that your queue prompts inform the caller that he can elect to leave a message pressing 1 at any time, and that in your queues.conf file you have the following entry for your queue: context = foo Now, in extensions.conf (or whatever included file you're using) you'll need something like: [foo] exten = 1,1,VoiceMail(1234); 1234 being the voicemail box you've designated to handle these calls exten = 1,2,Hangup -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.3 - Release Date: 05/03/2005 ___ 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 good SIP receptionist phone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Saturday, April 30, 2005 12:53 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A good SIP receptionist phone In a multi-tenant environment, is there a way to display, on the phone, which DID (which tenant) is being called? Yes. We've done this by simply prepending a meaningful string onto the front of the CIDName. It's a total kludge, but it works. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 04/29/2005 ___ 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] Confused on G723 and G729
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, April 28, 2005 8:31 AM To: Adam Goryachev Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Confused on G723 and G729 I'll gladly pay $10 a license... I'm all for supporting digium... however, I was under the impression that there was also some huge one time fee of like $2,000 or something. I guess I was wrong... ok now bad.. So I purchase the license from digium... then what happens/what needs to be done on Asterisk? Be aware that the license fee is $10 per instance. Each leg that is transcoded to or from G.729 on your box will use one license. So if you want to support 20 simultaneous callers checking their voicemail, you'll need 20 licenses. The installation process is well documented on Digium's web site. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.4 - Release Date: 04/27/2005 ___ 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] Confused on G723 and G729
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, April 27, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Confused on G723 and G729 My question is.. if my voip terminator supports G723 and G729 only, do I still need a license? Or is that considered pass-through? If so, do I need to do anything special to get it to work? It is pass-through if both end points are using G.729. You need a G.729 license for every instance where a G.729 stream is encoded or decoded on your box. If you connect G.729 endpoints together, this isn't happening so no license is needed. Same goes for G.723. I'm also a litle confused about why G723 can do pass-through but can't do voicemail access? There is no G.723 license available for asterisk, ergo no way to transcode the voicemail and other promts into that format. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 04/25/2005 ___ 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] VOIP Gateways Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Tuesday, April 26, 2005 12:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VOIP Gateways Asterisk We are planning to do an * install in an apartment building, this building is going to require somewhere in the vacinity of 20 E1 lines (each with 30 voice channels). Short of buying 20 Servers with Digium cards, what are my options in having the E1 lines terminate on some other hardware and then having the calls passed through to Asterisk to perform the PBX type functionality ? First of all, one has to ask why you need PBX functionality at all? As this is an apartment building, won't the tennants be looking for PSTN dial-tone, rather than local (PBX) dial-tone? Assuming that there is something missing here... You could treat this as a hospitality (hotel) type environment, but would certainly have to allow for a lower station:trunk ratio. Still, the thing to do would be to deploy channel banks (Adtran, Adit, etc.) that would be aggregated into the number of E1 ports on your * box required to handle the projected traffic. In other words, you will not encounter a condition where every station will be in use at the same time, so you don't need that many DS0's (channels, if you will) at the PBX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.3 - Release Date: 04/25/2005 ___ 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] Unbelievable...
-Original Message- [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unbelievable... Having worked as a senior manager in a technical organization, a large number of tehcnical people simply do not comprehend some words (or read other words into whatever they happen to be reading), or, jump to conclusions based on their technical knowledge that are unreasonable (contractually or otherwise). The wording is very obviously oriented toward those types, and I'd bet a fair amount they _still_ receive calls that are clearly answered on their web site. I'm sure this is true. Users, which is to say CUSTOMERS can be maddeningly clueless at times. However that is still no excuse for bullying and threatening. Qwest and others have learned over the last several years, and much to their dismay, that even simple indifference to customer concerns will result in a wholesale exodus as soon as other alternatives become available. Treating customers with the outright contempt that LiveVoIP displays with the statement in question is, again, staggering in it's short-sightedness. Regardless of what their web site says, they've provided me with the best service of the half dozen itsp's that I've worked with directly. And, I don't work for them or represent them. My experience with them has been likewise positive, which proves that they are at least capable of providing good service, on occasion. The fact that some users are frustrated to the point of posting here in this list in order to get the attention of the company's principals, SHOULD strike those principals as a clanging alarm that something in their customer service system is broken. Sadly, the lessons of Customer Service 101 appear to have been lost on them. And that's a shame, because as we both know, they are doing a largely good job, and it is in everyone's interest (theirs and their cusomters', at least) that they continue to do so. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.16 - Release Date: 04/18/2005 ___ 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] Unbelievable...
Unbelieavable, and utterly disgraceful. Anyone found responsible for establishing such a policy would quickly find their ass on the street in any organization that understands the first thing about customer service. One doesn't build or protect a business by threatening and bullying one's customers. If one is worried about the bad impression that complainers are giving about the operation, figure out WHY they are driven to such extremes and DO SOMETHING ABOUT IT. It isn't rocket surgery. The principles of running an effective customer service organization are well known and readily available to anyone. The mind boggles... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Sunday, April 17, 2005 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unbelievable... Sure sounds like a veiled threat to me. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.15 - Release Date: 04/16/2005 ___ 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] large analog to asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shane fowler Sent: Friday, April 15, 2005 10:10 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] large analog to asterisk 600 analog connections. Some rooms have 2-3 phones so as a rough number i'm saying 700 total. I see where some people use the Adit 600 to do up to 48 analog connections that trunks over 2 T1 connections back to asterisk but for 700 phones thats 15 Adits with 30 T1'show in the world would you do that?? just several asterisk servers with 2-3 Adits per server? is there any other way? I'm open to suggestions. Remember that in a hospitality environment, the volume of simultaneous calls is typically quite low, given the number of stations in the system. You could use 600's with the CMG-02 cards to backhaul to asterisk via MGCP. Asterisk's MGCP handling is not as robust as it might be, but it may serve your needs. Another option would be to bank on that high stations:calls ratio. In other words, you'll never need to provide 700 DS0's directly into the PBX. We spec'd a very similar (400 stations) hospitality system recently using a slug of Adtran 624's hanging off of an Adtran 830 equipped with 5 quad T1/PRI cards. Careful planning and dial-plan design can keep most inter-station traffic at the 830, with only those calls requiring trunk or PBX feature access traversing a small number of T1's between the 830 and the PBX (asterisk). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.11 - Release Date: 04/14/2005 ___ 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] Low cost box for hosting Asterisk and at least oneTDM400p
Chuck Bunn Sent: Monday, April 11, 2005 9:38 AM To: Linux - PBX, Asterisk Subject: [Asterisk-Users] Low cost box for hosting Asterisk and at least oneTDM400p Can anyone recommend a very low cost box that could support Asterisk and at least one (preferably two) TDM400p cards and cost less that $150 (preferably under $100). The short answer would be no, at least not with new parts. You will be hard pressed to get a suitable mobo, CPU, RAM, NIC, HD, and case for anywhere near $150. That said, the second-hand and surplus markets are probably an excellent source for systems meeting your specs, but the only recommendation one could possibly make is see what's out there or Ebay. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 04/07/2005 ___ 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] Linksys PAP2 Dual Incoming Calls
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 11, 2005 3:37 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Linksys PAP2 Dual Incoming Calls Hi List, Im facing a strange problem using a linksys-pap2 (two ports) ATA: I cant have two simultaneous incoming calls when i use g729 codec, if i use g711 (alaw) there is no problem, is this a know issue or am i missing something? Known issue. Apparently, the PAP2's CPU doesn't have the horsepower to do two G.729 calls at once. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 04/07/2005 ___ 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] Detecting Downed SIP Phone
John Goerzen Sent: Monday, April 04, 2005 6:06 PM Subject: [Asterisk-Users] Detecting Downed SIP Phone I recently encountered an odd situation: the network cable to my SPA-841 got unplugged while it was in the midst of a call. I got it re-plugged in about 30 seconds, and the phone rebooted. The phone showed no evidence of the previous call in progress and worked like normal. Asterisk, on the other hand, believed the call was still in progress -- my outgoing line was still in use, and it showed up in the show channels list. I resorted to the soft hangup command to terminate it. What could I do so that Asterisk would automatically terminate a call in these situations? Check out: http://www.voip-info.org/wiki-Asterisk+sip+rtptimeout -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 04/01/2005 ___ 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] Asterisk - Altigen
-Original Message- [mailto:[EMAIL PROTECTED] On Behalf Of Dan Perik Subject: [Asterisk-Users] Asterisk - Altigen Has anyone successfully tied together an Altigen system to an Asterisk system using VoIP (ie. not using hardware (FXO/FXS cards, etc.))? My experience with the Altigen's IP stack is a bit dated, so take this for what it's worth... At the time I was working with it, their VOIP implementation was so bad, that we abandoned it, and resorted to connecting spare analog ports to a Multi-Tech VOIP gateway. This solution worked like a champ. Even if Altigen's VOIP implemenation has gotten more solid, I'd recommend against using it, if for no other reason than the fact that it uses H.323. The H.323 support in Asterisk is spotty. In certain configurations, it seems to work fine, but others, H.323 -- SIP, for example, it seems to have issues. If you have much time to spare, and you already have the VOIP licenses for the Altigen, I guess you've got nothing to lose, but I wouldn't try it under any other terms. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 04/01/2005 ___ 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] Webmin
Don't. It is broken. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Thursday, March 31, 2005 3:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Webmin How do I install the asterisk module for webmin? ___ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 03/29/2005 ___ 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] Problem compiling asterisk-addons
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, March 23, 2005 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem compiling asterisk-addons Eric wrote: Hi, I am getting an error trying to compile the asterisk addons: cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make: *** [cdr_addon_mysql.o] Error 1 Can anyone suggest something I could try? Are you actually installed asterisk? Do you have /usr/include/asterisk/asterisk.h? I seem to recall chasing this one before... Eric, check to make sure that you have the mysql libraries (mysql-dev package) installed. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.4 - Release Date: 03/18/2005 ___ 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 configuration changes for Outbound
Doing this with no notification whatsoever, let alone notification sufficiently in advance of these changes, was stupid and careless. This move probably broke a significant number of your customers' telephones service. One can only guess at the impact that this careless move had on your customer service department. In the future, give some thought to planning such changes more carefully, announcing them well in advance of implemenatation. I am satisfied enough with my BroadVoice service that I will overlook this incident, but there are lots of other vendors out there. Surely, at least one of them has more concern for their customers than BroadVoice has demonstrated with this fiasco. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Weber Sent: Saturday, March 05, 2005 9:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BroadVoice configuration changes for Outbound Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 03/02/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 03/02/2005 ___ 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] E100P to Valiant E1-PRI GSM gateway
Looking for zaptel/zapata configuration parameters to successfully communicate with a Valiant GSM gateway as above. Surely someone has done this? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 02/22/2005 ___ 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] High capacity voicemail
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of izo Sent: Thursday, February 24, 2005 5:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] High capacity voicemail Hi, Does anybody has experience with high capacity PSTN voicemail and asterisk, running more then 5k mailboxes for PSTN users ? How many mailboxes can I serve with 4xE1 card if we assume that we have enough harddrive capacity. What would be server requirements. Would the CPU load be the same when storing voicemails in gsm format as compresing to gsm for ip calls ? Any hints would be greatly appreciated Given the hardware requirements documented here: http://www.digium.com/index.php?menu=faq#General_10 you'd probably want a very stout dual Xeon machine. Forget using ATA hard drives. You'll want to be shopping for a storage solution that has as little impact on CPU resources as possible. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 02/22/2005 ___ 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
