Re: [asterisk-users] Asterisk desktop tools for OS X
Yaah!!! Mac! I am a big user of OS X. Can't help it. Macs eye candy draws me in like my wofe. :) And.. I've never had a single issue with it. I also host virtual Ubuntu, Red Hat and XP :( on the same box using VMware. Sorry about the Mac rant. Just glad to see some Mac / Asterisk attention... I have multiple Asterisk servers in place and would REALLY be interested in your tool set. I can test it on Leopard or Tiger as I have both in available. Thanks, Jim - Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing time-out
Ok, probably a dumb question. I believe I already I know the answer, but thought I would get feedback from others. One of the issues with user devices at the end Asterisk is dialing time out. This is a parameter within each hardware device. So if I set it to 3 seconds it appears from the moment after going off hook any key press starts a timer allowing me 3 seconds to enter the next number before Asterisk times out and generically says I'm am sorry that is not a valid extension. Now this is ok, of sorts. The fault in this is when you dial a valid number you are stuck waiting 3 seconds for the system to out pulse and connect. This clearly separates Asterisk from the traditional TDM platform behavior where a time out can be REAL LONG allowed people to dial at a snail's rate without upsetting the phone system but then immediately out pulsing when a number match is met, regardless if the number match is a 4 digit extension or 7 digit phone number. Is this one of the reasons and purposes Asterisk has a real-time option? Thanks, Jim ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
We are in need of an IAX based hard phone. We have used softphones and USB headsets already and they are greatly affected by the other software running on the Windooz laptops and PCs of our users. Does anyone know where we can go to find IAX based hard phones in the US? The one on this link looks very nice. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Administrator TOOTAI Sent: Tuesday, November 06, 2007 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mystery phone! Kyle Sexton a écrit : Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts, SIP and IAX able, nice audio Good product. -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have used the Grandstream GPX2000, HT503 and GXW4104 gateways. Quality is in all cases are on the lower end. The quality I refer to is buggy software and poor call quality. I have been involved with Telecom since the early 80s and dealt with a lot of phone systems. The Grandstream phones just plain feel cheap. Real Walmart quality, not professional business class equipment. The phone functioned ok and was super easy to setup but complaints of echo and poor volume levels were common. They may be better as we have not used them in over 6 months. We have recently used their gateways due to good pricing and their economics fit our solution base well but ran into issues with them. I believe their gateways will get improved as both are new and on early firmware releases. However, we got upset with poor support. Either no call back at all or a useless email a day later with little to no information to help solve our issue. In Grandstream's defense it may be we are just too small to matter and that's ok. We prefer to go elsewhere and deliver product that when the average user picks it up to talk on it they say this is quality stuff. Asterisk is as talented as the firm that programs it BUT the phone is crucial in the end user's system satisfaction. Regardless of what you put in the back room the phone IS the device that sets the impression to your client if you are delivering a quality solution. We would do Cisco because it is high quality but we don't care to fight with the configuration or licensing issues. We would do Polycom, and probably will, but have not had the time to jump to through the hoops needed to acquire good enough pricing to make money selling them. We feel Aastra is a good compromise in delivering quality product to make the customer happy with their decision while still making us to make some sort of small profit for our time. It's solid and provides a quality feel and function. This said, Grandstream is not junk and this is not meant to be a Grandstream rant. I would like to apologize if I jumped in too quick sounding that way. Grandstream is just the lower end of quality and should be deployed in applications where the client is willing to accept that. That's not our marketplace. If you want easy to configure, low cost, slam dunk Asterisk deployments then Grandstream works. But the end result will not be as good if you build a system with Cisco, Polycom, Snom, or Aastra. We've even tested Avaya 46XX phones on Asterisk. They sound GREAT! Probably one of the best. We just can't get Asterisk to light the messaging waiting light on the phone. Arrggg! You need to decide what your marketplace offering is and what your clients are willing to accept. If call quality is the most important then our testing shows nobody beats Polycom or Avaya. Someday we are going to beat the Avaya message waiting light issue. If quality of deskset feel is the most important factor them Avaya and Cisco stand out as best. We will not put configuration into a factor simply because the customer uses the tool we are expected to configure it to their needs. We won't sell them any device based on it being easier for us to configure. I would like to hear what people say about Snom as their sets look very nice. Sorry for the novel, all I really wanted to express is Grandstream is cheap, look before you jump. Good luck on your decision... Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard
RE: [Asterisk-Users] FreePBX in production?
Since FreePBX is module based it seems that with all the good people out on the internet there is someone will write an add-on to extend the capabilities for those that need it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, May 01, 2006 6:52 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX in production? Wouldn't use it in production for a customer personally. Too many limitations in terms of having a flexible diaplan. What would be nice though is if they were to produce a 'lite' version that gave a gui interface to add/change/move things - sip.conf, voicemail.conf, meetme.conf but staying well away from extensions.conf Craig - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users-List asterisk-users@lists.digium.com Sent: Monday, May 01, 2006 5:19 AM Subject: [Asterisk-Users] FreePBX in production? Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near usable even with the latest beta1 code. Is it just me or what? Rich ___ --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 ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] USB conference phone
I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operatesnice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] I am looking for a webphone on MY SITE
I need the same exact thing. Our site is almost all Perl with a little PHP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 26, 2006 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITE I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] I am looking for a webphone on MY SITE
Are you looking for something that visitors to your website can use to call you? This is what I'm looking for. Basically a on-screen phone with "push to talk" buttons that are directed into a department queue. I'm open to any suggestions. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom HaydenSent: Wednesday, April 26, 2006 9:40 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. --Tom On 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThis e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] I am looking for a webphone on MY SITE
That's basically what I'm looking for but wondered if we could do it in Perl. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: Wednesday, April 26, 2006 10:07 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE The only one I have heard of is WebIAXhttp://www.voip-info.org/wiki/view/WebIAX On 4/26/06, Tom Hayden [EMAIL PROTECTED] wrote: What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. -- Tom On 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--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 This e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] USB conference phone
Personal preference. I'm not a big headset guy. The real point of my reply was to say how impressed I am with USB talk quality when compared to ahardphone on Asterisk or our Avaya Communications Manager. Like my wife says, I guess I'm not being clear... :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Wednesday, April 26, 2006 10:24 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] USB conference phone Kerry, do you actually own one? Have you used it for long? What are you using it for? (jim personally I cant see the point of using your phone when I have a very good quality headset and mic.). Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry GarrisonSent: Wednesday, 26 April 2006 10:36 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB conference phone This is an excellent USB speakerphone http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim HouserSent: Wednesday, April 26, 2006 6:26 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB conference phone I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operatesnice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] T1 to cross connect remote PBX and asterisk
I have our Avaya connected to Asterisk using NI D channel protocol over a standard ESF/B8ZS span. It works great. Pretty easy. On Asterisk's side I just had to tell it: in zapata.conf: [channels]switchtype=nationalsignalling=pri_cpegroup=1channel = 1-23 in zaptel.conf: loadzone= usdefaultzone= usspan=1,0,0,esf,b8zsbchan=1-23dchan=24 Jim From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: Tuesday, April 18, 2006 11:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] T1 to cross connect remote PBX and asterisk Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the technical details on how to set up asterisk or the remote PBX, just need a sanity check on the idea of using the PTP T1 as a cross connect facility. If they were local to each other I would simply drop in a T1 crossover cable, but they are not J Thanks! This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] T1 to cross connect remote PBX and asterisk
Daaah, you are correct. A typo on my part, not a cut paste from my actual build. Make that span=1,1,0,esf,b8zs -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, April 18, 2006 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 to cross connect remote PBX and asterisk Jim Houser wrote: I have our Avaya connected to Asterisk using NI D channel protocol over a standard ESF/B8ZS span. It works great. span=1,0,0,esf,b8zs Shouldn't you be getting your timing from the Avaya? Doug ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Building Asterisk embedded device
http://gumstix.com/waysmalls.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sam Sent: Friday, March 31, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Building Asterisk embedded device Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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, QSIG and Tenovis PBX?
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it working great as PRI. Am I wrong about the Q.SIG support? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Sent: Friday, March 31, 2006 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX? Hello Dinesh I got a Panasonic KX-TDA100, can you tell me please how can you configure the PBX side? Qsig slave? master? and the other side of the asterisk? I got TE100P Regards, Daniel Dinesh Nair wrote: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Marketing Materials
Digium.com has pdf brochures on Asterisk and their hardware you can download. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: Wednesday, March 29, 2006 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Marketing Materials The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] FreePBX AAH
I wanted the user interface of FreePBX over what is provided in the latest version of AAH. I installed the latest version of AAH and then just installed FreePBX over the top. It went fantastic and I do like the FreePBX web interface better than the latestAAH. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard AmermanSent: Wednesday, March 29, 2006 3:32 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] FreePBX AAH This question confuses me. My understanding is that FreePBX is just AMP renamed and AAH comes with AMP setup as the primary way to manage it. So, is the question realy that the user wants a newer version of AMP (read FreePBX) then the one that comes either with the newest version of AAH or the version that they have installed? Richard On 3/29/06, Dovid Bender [EMAIL PROTECTED] wrote: snip wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience... /snip the only way to learn is by playing. a little over a year ago i knew nothing about linux. google. is your friend. New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 Tools for OSX
Yaah! I'm a Mac fan. PPC mini in my home office and Intel dual core mini in our audio video room. I'm a fan of JackenIAX softphone and look forward to any OS X integration with Asterisk. Thanks and keep us posted. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devraj Mukherjee Sent: Tuesday, March 28, 2006 5:05 PM To: Asterisk Users Subject: [Asterisk-Users] Asterisk Tools for OSX Hello Asterisk Users, I am an Objective-C enthusiast and have been writing some clever tools to integrate Asterisk functionality with Mac OS X applications. Please find my project on http://www.sf.net/projects/astrxtools4osx/ The objectives of my project are as follows 1. Implement an Objective-C framework to communicate effectively with the Asterisk Management Interface 2. Address Book plugin to enable call back functionality 3. A System Preferences pane to allow administrators to easily configure Asterisk options on a Mac 4. Dashboard Widget that allows users to quickly call arbitary numbers 5. iTunes integration to stop and star iTunes to play when the phone rings etc. The source code is in pre-Alpha stage at the moment but I am hoping to release a Beta at the end of next week. Please feel free to download and use these extensions. I hope they turn out to be useful and would appreciate any feedback. Devraj ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreePBX AAH
Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] FreePBX AAH
My understanding is you can install it on any Linux server running Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, March 27, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX AAH Pardon the question, but what I understand of FreePBX is that it's basically Asterisk with a web interface and some additional modules. Is that correct? Can you install FreePBX on a system which ALREADY has asterisk up and running or does it require ITS version of asterisk? Thanks, Waldo On Mar 27, 2006, at 12:29 PM, Tom Vile wrote: Yes, you can. On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote: Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] problems with emailing voicemail
Title: Message I'm not real knowledgably in Linux, but have you loaded Webmin so you canlook at the status andmessages in sendmail? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivudeSent: Friday, March 17, 2006 9:06 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] problems with emailing voicemail Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get Asterisk going, I'm hoping someone can steer me in the right direction! Any suggestions where/how to troubleshoot? Many Thanks, Hugh This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Feedback from VON expo! Info on * HA and Polycomphone!!
Gabe. Who was the call-center program from? Thanks, Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Thursday, March 16, 2006 2:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! Hey group, I just got back from the VON expo. It was insanethere were so many companies there. The #1 thing ***EVERY*** company focused on was convergance - getting all your communication devices to intergrate with eachother. There were some nifty products out there that did some cool stuff :-) Of course Digium/Asterisk was there and I had a list of questions for them. I went by several times asking more and more questions...by the last visit, these guys were running from me because I was driving them nuts :-) Here are all the questions I asked them (this is not word for word...just a summary): Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. Q: What about failover without losing a call A: IBM has been able to make asterisk do this. However, at this time we are not working on any solution to offer this as part of the program. Q: Do you plan on offering support for other distros for Asterisk Business Edition? A: [uncertain answer] Not really sure...maybe SuSE...not sure Q: When is asterisk going to fully support video? A: Asterisk can complety support video using H.261, H.263 and we recently added support for H.264 Q: What do you recommend as the best solution for HA? I got two different answers for this from two different people there. Both made good sense and are basically what everyone is doing now. Here both approaces are in a nut-shell: Approach 1 (seemed to be the preferred method): Use DNS-SRV lookups for all registrations. This will distribute the calls among the * servers. Next, you configure your servers using regexten and DUNDi. You use regexten to dynamically create the exten = 1234,1,NoOp when a phone registers with that server. Then when a call comes in, you use DUNDi to try to complete the call locally. If the phone is not registered to that server, then do a DUNDi lookup to find the server that the phone is registered to and then pass the call over IAX to that server to take it to the phone. Of course the phones will need to have a short registration expiration so they update frequently because if the server they are registered to crashes, until it re-registered, no server can access it. But by doing this, the phone will re-register to another server and then the next DUNDi lookup will then go to this new server. I asked about the load of having many phones registering frequently and he said it is no big deal at all. He also said it was very important to make sure cache is disabled in DUNDi!!! Each call that is made should result in a new query. This will ensure the calls are not getting old cached info which may no longer be accurate. Approach 2: Use a SER box to handle all registrations. The SER box will take care of distributing the load between the * boxes. You do not use DUNDi or regexten in this case. Just let each * box function on its own. If one of the servers fails, SER will not use it to terminate calls. Sinces the phones are registering to SER, and all incoming calls will be routed to SER, you do not need to worry much about the * boxes. You just need to make sure you have your SER boxes in a cluster to fail-over in the event of failure. Overall theme of the Asterisk stand: selling third-party products. In the there section, Digium had 10 seperate vendors that have teamed with them to sell special programs/products/services that intergrate with Asterisk. One was a call-center program, another was a resellers package, another delt with firewalls and NAT, another for voice recognition, another was Intel (that has partnered with Digium to offer drivers in the ABE for the intel cards), another was some email, fax, chat, presence, etc. kind of box that sits in front of * to combine all these servicesand some others I dont remember. It felt like I was walking into an infomercial! I also spoke with Polycom guys a great deal and asked many questions: Q: Do you plan on offering 10/100/1000 ports on the phones? A: Yes, in the near future Q: Do you plan on offering a standard phone jack for failover purposes? A: No, we have no talks of this. However, I will take this idea to the production development team. Q: What is the services button ever used for? A: This is only operable in the 601 and is used to launch the XML browser. We have partned with many companies to offer you sports, weather, stock, movie ticket info...etc that can be fed directly to the phones screen. Q:
