http://blog.gmane.org/gmane.comp.telephony.pbx.asterisk.biz hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32916 <pre>We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the telecom industry, installing, configuring and maintaining PBXs. A couple of weeks ago he moved into a new role (for him and for Digium), Community Support Manager. In this role he'll be the primary person responsible for ensuring that Digium's community services are providing what the community members need, that the systems are operating properly, and that issues and questions are getting the attention they deserve. He'll be working closely with our Community Director as well, especially for events like AstriCon and others. He works directly with the software development team at Digium, which will allow him to focus almost exclusively on technical issues and discussions. We're quite excited that he has taken on this role and we expect that you will soon see the benefits of his activities across the community! </pre> Kevin P. Fleming 2012-05-25T14:41:12 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32915 <pre>On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight Time, GMT-5), the servers that Digium uses to provide many services to the Asterisk community will be relocated. This will mean that these services will be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org -- _____________________________________________________________________ </pre> Asterisk Development Team 2012-05-23T14:45:00 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32905 <pre>Hi, We want an option for sip.conf that would be similar to qualify. We need to know if the peer is responding to OPTIONS packets or not. This way if it is down we know right away to continue in the dial plan. The issue with qualify is that if the response time is 250 MS and when Asterisk sends an invite it does not get a response in 250 MS then it sends the invite again which then "irritates" some gateways. Regards, Dovid -- _____________________________________________________________________ </pre> Dovid Bender 2012-05-23T06:14:28 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32903 <pre>Hi All, For any who might be interested in this sort of work, we have a project coming up in the near future for which we will require a call-center in one country and trunk-scale connections between it and two and later three other countries, all in Asia. The end users will call into GSM SIMs hosted by Asterisk servers situated in their native country, and then SIP will handle the international relay between those hubs and the call-center staff at our primary location; who themselves will also have GSM SIMs to receive direct calls from domestic clients. The call center's dialplan will need to be arranged into five different groups to represent the five different companies that will all be working under the same umbrella. The current numbers in terms of SIM Cards/GSM Channels &amp; Day Operators &amp; Night Operators are: 18 &amp; 15 &amp; 6, 14 &amp; 8 &amp; 3, 34 &amp; 4 &amp; 1, 21 &amp; 7 &amp; 3, 4 &amp; 3 &amp; 1. So a total of about 91 channels and 51 operators to start, with rapid growth expected for at least two years after the first 6 months. There are three different mobile network operators serving the call-center site, and a similar variety in numbers at the first three other countries. We need all the usual bells and whistles: ACD, Call Recording, Queue Monitoring &amp; Statistics, ability to pause out of queue for short breaks, Agent Metrics (very important), MoH, blind transfers, but not voicemail or call parking. Agents will use soft-phones to start but we're prepared to move to real phones if quality demands it. No need for screen-pops or desktop computer integration yet. One other big item: QoS on the LAN — we have none as of yet, are currently using a Debian box as our router and unmanaged gigabit switches, so we'll definitely need this added. I don't need a formal proposal yet, if you're interested then please just send your qualifications and a rough ball-park estimate on time and *labor* costs but don't stress on them, they're not something you'd ever be held to: I will narrow the pool of applicants down to a short-list and then ask for more detailed, realistic figures as part of what would be considered an official proposal. Thanks in advance! Jonathan CTO IntelligentMillionaire.com -- _____________________________________________________________________ </pre> Jonathan Barratt 2012-05-22T18:31:52 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32902 <pre>i ordered an iphone4s and mac from this eshop , 20% Off all orders in May. now i had recive it , i like it very much so i tell you , hope you can try too take a look :*&lt;depthdeals.com&gt;* regards </pre> Alexander Argov 2012-05-20T02:57:44 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32901 <pre>I am interesting in hearing about experiences with Independent RespOrgs --Don Don Kelly PCF Corp People Come First 651 842-1000 -- _____________________________________________________________________ </pre> Don Kelly 2012-05-17T16:04:27 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32893 <pre>Hi, does anyone know where I can get an Israel "kosher" DID for a calling card? (https://en.wikipedia.org/wiki/Telephone_numbers_in_Israel#Kosher_Numbers) Thanks, -Avi Marcus BestFone -- _____________________________________________________________________ </pre> Avi Marcus 2012-05-13T16:54:54 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32889 <pre>hello folks anyone know of a carrier that can offer a SMS enabled did's in north america including the US and canada ? SMPP or sip delivery is required thank you Meftah Tayeb IT Consulting http://www.tmvoip.com/ phone: +21321656139 Mobile: +213660347746 __________ Information from ESET NOD32 Antivirus, version of virus signature database 6830 (20120126) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _____________________________________________________________________ </pre> Josef Grand 2012-05-10T12:37:36 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32883 <pre>Does anyone know of a SMS provider where a 10 digit caller ID can be sent? Im not even sure if this legal and have not been able to find anything on it. Any comments would be great! Thanks -- _____________________________________________________________________ </pre> Brad Bendy 2012-05-10T16:53:53 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32878 <pre>LES.NET offers Canadian DIDs with 96+ Channels Delivery by SIP to your conference bridge, G.711 + G.729. E-mail sales&lt; at &gt;les.net, or Phone 1-888-399-VOIP xSales Les -- _____________________________________________________________________ </pre> LES.NET (1996) INC. 2012-05-09T03:49:35 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32877 <pre>Hi, I am looking for developer who would like to fix this bug: https://issues.asterisk.org/jira/browse/ASTERISK-19761 Please email your offers to chris (at) wima ( dot ) co ( dot ) uk Regards, Chris -- _____________________________________________________________________ </pre> Chris Maciejewski 2012-05-04T15:27:00 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32876 <pre>Dear List members, We would like to share information about AQuA product we developed for VoIP community that is used in Asterisk based voice quality monitoring system we provide to perform end-to-end voice quality testing that does not rely on traditional VoIP parameters like packet loss, jitter, latency. Even now and then we get questions on how AQuA is similar / different to PESQ. Well, AQuA surely differs from PESQ, because it’s using different perceptual model, returns different parameters (AQuA returns prediction for objective MOS score according to P.800 and PESQ values are from 1 to 4.5 considering that nobody can distinguish MOS 4.5 and MOS 5), has different features, provides more information on reasons for audio quality loss. We clearly state that AQuA is not PESQ. There are cases which PESQ fails to detect, but AQuA does catch the degradation, read this article Perceptual Evaluation of Speech Quality (PESQ) (http://www dot microtronix dot ca/pesq.html), and there are reports that state PESQ may not give accurate scores for example in mobile and VoIP end-to-end testing (you can google for “Limitations of PESQ”). However, the point is not that AQuA is better than PESQ, it’s just different, but different in many ways. We created AQuA to provide VoIP community with inexpensive tool to measure and monitor voice quality in VoIP, mobile, PSTN and converged networks. Currently we have customers that use AQuA for all types of networks including sattelite communications and they are quite happy, because the main point is that one needs to know whether the quality is good or bad and if a tool delivers this information at a certain precision then it’s useful. When evaluating voice quality in VoIP or other communication network our customers report two main issues: 1. test signal may have different level (RMS amplitude) than the original, and in this case it requires amplitude adjustment that is done by free SOX utility in AQuA 5.x. AQuA 6.x that is ready for release has built in feature to adjust RMS amplitude using our in-house algorithm. 2. test signal may have “silence” in the beginning and / or end of the recorded audio file. In order to synchronize two signals in time domain one can use SOX as well, or use -trim parameter in AQuA, or even both. There are a lot of parameters one can adjust in AQuA, but in most of the cases they are set to default values and we deliver batch scripts for Windows and Linux with pre-set parameters. Typical test then looks like Windows: test.bat reference.wav test.wav Linux: ./test.sh ./reference.wav ./test.wav To summarize AQuA and PESQ do the same thing differently. If one requires to provide its end customers voice quality score accordoing to International standard, then it’s surely PESQ, because it’s a recommendation from ITU-T P.862/P.863 bla bla bla that costs a lot of money and is provided by different companies as a software / hardware combination, boards, features, bla bla bla… And if you are a VoIP service provider or a mobile operator or DSP algorithms developer then AQuA is your best choice – no royalties, no specific hardware (you want a dialer connected to E1/T1, GSM or sattelite network? then use Asterisk VQM – open source based voice quality monitoring soluton) – just pure voice quality assessment. If you have questions you are always welcome to contact us! Best regards, Sevana Oy http://blog.sevana.fi/aqua-vs-pesq/ AQuA page: http://www.sevana.fi/voice_quality_testing_measurement_analysis.php -- _____________________________________________________________________ </pre> sales< at >sevana.fi 2012-04-24T18:00:24 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32875 <pre>Folks, We're looking to add some linux / asterisk / voip / foip talent to our bench here in Philadelphia. I'll spare the list the trouble of reading a long job description, but basically we're looking for a fairly hard-core linux nerd (I use that term in the most favorable sense, being one myself) who might enjoy troubleshooting a wide variety of weird and wonderful customer issues associated with voice over IP and fax over IP hardware and software. This is a junior position staffing the front line, our first level of escalation, with significant room for growth for the qualified candidate. The position would involve speaking to customers by telephone, so some social skills will be necessary ;-) This is NOT a telecommuting position. If you'd like to know more, email work at ifax.com and we can send along a more complete job description. -Darren PS - not sure how many people actually subscribe to asterisk-biz, but I figured this would not meet the posting guidelines for asterisk-users. If I'm wrong there, perhaps a list administrator could let me know by private email and I'll repost to -users for (presumably) wider reach. -- _____________________________________________________________________ </pre> Darren Nickerson 2012-04-24T16:55:45 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32872 <pre>Hi, Does anyone know a VSP that offers Kenya DIDs? Hopefully they would have a colo relatively near (europe). Thanks all, Jen -- _____________________________________________________________________ </pre> Jennifer Bartlett 2012-04-23T16:30:20 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32866 <pre>Hi - the best prices I've seen for canada DIDs are $1 + .01 per minute or $4.95 for 2 channels via voip.ms Even localphone wants $3 for a 2 channel Canada DID. This pricing is prohibitive for a calling card and offering competitively priced international forwarding... If you've got better pricing without large minimums, please inform. -Avi Marcus BestFone -- _____________________________________________________________________ </pre> Avi Marcus 2012-04-17T21:37:31 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32859 <pre>Anyone have any good termination rates to Barbados Digicell? Not looking for wholesale - just retail client Sent from my iPhone 4S -- _____________________________________________________________________ </pre> Robert-IPhone 2012-04-16T00:56:17 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32857 <pre> Looking for someone to help clean up and give some configuration advice on my current dial plan. This is just your standard typical toll free IVR with a few unique features, for a very small start up, that will be using Asterisk for its telephony needs. Also, got a few educational questions regarding some of the internals and some best practices type questions. Thanks,Peter -- _____________________________________________________________________ </pre> Peter M 2012-04-14T13:09:03 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32855 <pre>Friends, I've been running the Asterisk SIP Masterclass for many years now. It's time to run the last show - partly with new material. Compared with the very first Asterisk SIP Masterclass I would say that I've rewritten 90% of the material. That's what happens during the class. Students ask questions, you write a slide. The word changes, you write a slide. You realize you've been wrong, you delete or edit a slide. It's a moving target. The last one of these classes that I teach will be in Barcelona, Spain - June 11th to June 15th. * Why the last one? -------------------------- Things change and you need to follow. During the last couple of years I've been running many, many in-house trainings and workshops covering both Asterisk, SIP in general and Kamailio. There seems to be more demand for customized trainings that boost a team and help them move forward. I will continue with these trainings, as well as try to come up with other trainings that will run just a few times - more lab oriented possibly. * What is this class? --------------------------- From the sales material at http://www.avanzada7.com: "This class is focusing on building a scalable SIP realtime network. With a combination of theory and practical labs, you will learn how to setup and configure Asterisk and Kamailio - the Open Source SIP server - in a scalable enterprise or service provider network. We will go through various kinds of setups and give you insight in the design of real SIP networks with Asterisk running in enterprise and service provider networks. The teacher Olle Johansson, has many years of experience as an Asterisk developer as well as a community member of Kamailio.org. By spending a week with Olle, you will get a lot of insight into current and future features, bugs and implementation details in a way that's hard to get otherwise. 
 Olle is a consultant working with architecture and implementation of large scale communication platforms based on the SIP protocol. He has experience from service providers, universities, call center platforms as well as enterprise solutions. With experience of Unix and TCP/IP networking for over 20 years, he has a lot of insight and knowledge, which he is using as a teacher." The class is a five day high level class. You will meet not only myself, but also other students that work with these tools and protocols, learn from them and work together to solve issues in the labs. You need to have a basic knowledge of Linux (how to start/stop applications, edit text files and build applications) and Asterisk. This class is starting at a high level with Asterisk. If you rather use FreeSwitch but want to learn Kamailio that is no problem. You will just have to endure a few slides on Asterisk - but many of the issues apply to FreeSwitch as well as other PBXs too. The cost is 3.200 Euro ex VAT. Companies outside of EU do not pay VAT as well as companies in EU with a VAT registration number. If you have any questions or want to register, feel free to contact me directly. Looking forward to seeing you in Barcelona! Best regards, /Olle --- oej&lt; at &gt;edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! -- _____________________________________________________________________ </pre> Olle E. Johansson 2012-04-12T07:46:15 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32854 <pre>Hi, We have 2000 units Netgear WNR3500L open source Gigabit wireless N router for sale. Only $36.66 free shipping anywhere in USA (USA only). Add an USB drive (not included) into its USB port and you can use it as a mini Asterisk PBX platform. Please go to DD-WRT site for details if you need details how to install Asterisk in DD-WRT firmware. http://www.amazon.com/gp/product/B004UV4LOQ Yun -- _____________________________________________________________________ </pre> Sunny VoIP 2012-04-07T14:55:54 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32851 <pre> -- _____________________________________________________________________ </pre> Brian Meissner 2012-04-04T07:10:47 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32850 <pre>Bonjour, Tudor vous a invité à rejoindre Rencontres Francophones a Bucarest. Tudor dit : ---------------------------------------------------------------- Je vous invite. ---------------------------------------------------------------- Pour en savoir plus et nous rejoindre, cliquez ici : http://www.meetup.com/Rencontres-Francophones-a-Bucarest/t/if_1801102/?gj=ej4 Si vous n'êtes pas intéressé, n'effectuez aucune opération. Meetup ne conservera pas votre adresse. -- _____________________________________________________________________ </pre> Tudor 2012-04-02T19:39:41 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.biz