RE: [Asterisk-Users] open source sip softphone (Window OS version )
None of those are open source that I recall. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Thursday, June 15, 2006 6:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] open source sip softphone (Window OS version ) Asterisk guy wrote: are there any open source sip softphone (Window OS version )? ___ --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 xlite sjphone firefly (3rd party version) ___ --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] Voicemail to Email on Blackberry
Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] GXP-2000
With hundreds of installed phones now, here are my choices in order Linksys SPA-941/942 Polycom 501/601 Cisco 7960 Polycom 301 Snom 320/360 I would never ever ever sell a client on a SPA-841 or heaven forbid the GXP-2000. All the clients who bought those originally sold them off and went for better phones very quickly. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com Polycom 501 Linksys spa-941 Polycom 301 Sipura/Linksys spa-841 Grandstream GXP-2000 ___ --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
RE: [Asterisk-Users] TDM2400P with echo canceller not working
If you call Digium they will help you get the card configured properly. You get installation support with any of their hardware products. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Monday, May 29, 2006 12:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDM2400P with echo canceller not working Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found no info about how to use this card and how to correctly set zapata.conf, which zaptel version to use, etc... Does anybody knows how to use this card? TIA Giorgio Incantalupo ___ --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
RE: [Asterisk-Users] Web based interface
There are several listed at http://voip-info.org. For Management check out FreePBX, for recorded calls look for Asterisk Recording Interface. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne Sent: Saturday, May 27, 2006 9:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Web based interface hello is there a web based interface for IVR management, check voice mail, check recorded calls and ect. regards kani ___ --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
RE: [Asterisk-Users] features
To enable call waitng on extension 105 database put CW 105 ENABLE to disable call walting on extensions 105 database put CW 105 DISABLE From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled ChehabSent: Saturday, May 27, 2006 1:43 PMTo: asterisk-users@lists.digium.comSubject: Fw: [Asterisk-Users] features Dear In need to know how to add an asterisk feature as call waiting (*70) or disable call waiting (*71) for a user with out using a sip phone (DTMF)or gatewayI want to manage it form a command line or a Perl script or php web interface Or editing a file at asterisk directory . In need to it urgently Regards Thanks *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* ___ --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] Is NuFone Really Dead? Same as voipjet
This is the same with VoipJet, some people have good luck but my lines have been down for 3 months and all attempts at contacting them have gone unanswered. Hard to believe people still rave about their service. Here is a hint folks, if the company does not post a customer service PHONE NUMBER don't use them. Secondly, if they do have a phone number but nobody ever answers it, don't use them. Just because their email address is [EMAIL PROTECTED] doesnt mean its fast, or is even answered. It should be /dev/[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, May 25, 2006 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is NuFone Really Dead? On Wed, 2006-05-24 at 14:00 -0700, Andy Jefferson wrote: Went to their site today. Site claims they are still in biz. What is the story? What really happened to Nufone anyway? They say they are but I have 2 800 DIDs with them that are still offline, they stopped working more than a month ago and all attempts to contact support have been unanswered. I guess it is up to luck, some users have service others do not. Do you really want to work with a company like that? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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 virtualization
You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --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
RE: [Asterisk-Users] latest @Home questions
You will have far better luck asking this in the AAH forum or the FreePBX site. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Wednesday, May 24, 2006 1:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] latest @Home questions We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer sequence be ## rather than # I looked at the features.conf file and it didn't have an entry for blindxfer, so I added it. However, # is still the transfer character so it doesn't seem to be recognizing the settings from features.conf. 2. Not allow incoming calls to transfer at all. I've looked at the dial() string on incoming calls and they do not contain a t or T like I would expect for the channel to be able to transfer. As for the meetme conferences, the docs say that for all extensions defined, there is a meetme conference at 8ext. So extension 250 would have a conference at 8250. This isn't the case on our installation. I went to the Conference menu item and defined a conference at 1000 and I put an entry into the IVR for incoming callers to get into the conference and that works fine (except that after the PIN number, they cannot press # to signal the end of the PIN -- that will try a transfer). However, I don't have a way to get to the conference from an extension. I think I'm missing something, because meetme setup cannot be that difficult... Thanks for any help anyone can offer. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
RE: [Asterisk-Users] What and When is the next version of Asterisk?
1.2.8 would be the logical next version. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obelix Sent: Wednesday, May 24, 2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] What and When is the next version of Asterisk? What and When is the next version of Asterisk? /Obelix ___ --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
RE: [Asterisk-Users] US telco lingo
Could someone explain to a non-US dummy the following phrases I have seen on the list. I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout. We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6. What is 6/6? 2/3 devil? Normally I would take that to be minimum 6 second billiing and billed in 6 seconds increments. What is US48? Contentinal US, lower 48, all states by Alaska and Hawaii. What is blended? What you do with ice, alchohol, and a mixer ___ --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] Best VoIP provider for Asterisk
Depends on your location and your requirements. A generic post like this generally turns into a flame war. Please be MUCH more specific. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Tuesday, May 23, 2006 5:56 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Best VoIP provider for Asterisk Hi Friends,Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you.Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] What about T400 T1 cards?
Little personal preferance here (and hopefully some payback for some help you gave me a while back). My experience has been that unless you have echo cancelation on the hardware, the hardware isnt worth purchasing. This holds for the TDM and T cards right from Digium. If I am going to do a T1 install, I insist on the high end quad port card because it is the only one with hardware echo cancelation. Other good cards to look at are the Sangoma and the Rhino of which I have had good results from as well. But the low-end Digium and so far, any clone card, aren't worth the PCB they are printed on. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: Tuesday, May 23, 2006 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] What about T400 T1 cards? Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart ___ --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
[Asterisk-Users] Watchguard Firebox 1000 woes
We are trying to setup a sip connection behind a Watchguard Firebox 1000 and it is simply put...not working. The ports are all forwarded but the packets are not going out. It is as if the firewall simply ignores SIP packets. Has anyone seen this or have any idea what the issue could be? Watchguard so far has been of zero help. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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: Asttapi for Asterisk 1.2 Testers Needed (was RE: [Asterisk-Users]Asterisk TAPI - Outlook click2dial)
Please hook me up. I have customers dieing for something that works. All our systems are Asterisk 1.2.7.1 -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clint Sharp Sent: Tuesday, May 16, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Asttapi for Asterisk 1.2 Testers Needed (was RE: [Asterisk-Users]Asterisk TAPI - Outlook click2dial) I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to test this out, I'll send you a zipped copy of the TSP file (I haven't worked on doing an installer yet). I need to send out the debug build so I can generate information in case it doesn't work on anyone else's PC. Contact me off-list for copies. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Friday, May 12, 2006 1:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Oh.. :/ too bad.. I'll have to look at the source.. bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: Thursday, May 11, 2006 11:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Yes, I have the exact same problem. :( -Original Message- From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 5:48 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted - omniis.com. Version is 0.10 (newest) Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't even implemented in AstTAPI driver so that could be the reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T.S Sent: Thursday, May 11, 2006 1:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial I had similar problems when I first started to play with it. I've gotten Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But i don't know the version im using 0.0.8 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Wednesday, May 10, 2006 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation - once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --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 __ NOD32 1.1530 (20060510) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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 __ NOD32 1.1532 (20060511) Information __ This message was checked by NOD32 antivirus system.
RE: [Asterisk-Users] Asterisk with SIPconnect
We are a Cbeyond partner and have implemented their SIPConnect product. My main complaint is that they don't let me spoof the outbound caller id yet. They lock it down to one specific number. So if users with their own DID's want their number to go out for caller id, you cannot do that at this time. Otherwise it works great and we will continue to use it for future clients. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Gorby Sent: Monday, May 15, 2006 12:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with SIPconnect Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian ___ --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
RE: [Asterisk-Users] Voicemail WAV to PDA Problems
Our system is running all of the latest code and freepbx and would send the attachment to my MDA just fine and I was able to play it without any problem. My problem was that the MDA is a worthless turd and a complete joke as a phone. I took it back and switched to the backberry 8700g which has its own attachment problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Friday, May 12, 2006 9:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail WAV to PDA Problems Our asterisk server has been up and running for over a year and it works great. I have emails going to my account as an attachment and I can listen to them on the desktop and it works fine. I just got a T-Mobile MDA that runs Windows Pocket (or whatever they call it) and it can check email. If I have it download the email, it gets the attachment, but it can't seem to play it (it CAN play wav files). If I take the email that was sent to my home account and then forward it to myself and let the MDA pick it up, then it can play the attachment. So clearly it isn't an issue playing WAV's, or even WAV's from Asterisk, it's some email attachment issue with the way Asterisk or Postfix sends the attachment. Has anybody else run into this problem? If so, any help would be appreciated. Peder ___ --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
RE: [Asterisk-Users] How do I monitor the whole conversation on aZap channel ...
How are you trying to do it? ChanSpy or ZapBarge? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Azzopardi Sent: Wednesday, May 10, 2006 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I monitor the whole conversation on aZap channel ... I want to monitor the zap channel while the phone is answered by a human not by asterisk. Tzafrir Cohen wrote: On Wed, May 10, 2006 at 08:53:36AM +0200, Anthony Azzopardi wrote: How do I monitor the whole conversation on a Zap channel without answering it - the channel is hanging up, I think it's because it's not answered. If the channel is not answered, there is no (useful) audio in it to monitor. What exactly are you trying to do? -- Tzafrir ___ --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 -- Best regards, Anthony Azzopardi. Tel: 79713618 Email:[EMAIL PROTECTED] Sign up for free voip from http://line.sytes.net ___ --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
[Asterisk-Users] Linksys IP Device Bulk Provisioning Guide
I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_contenttask=viewid=73 Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] voipjet down?
I cant imagine anyone using voipjet as their only or main provider. And I'll say again, 3.9 cents for an ITSP is the most expensive I have found. Business grade termination is typically much less than that with top notch companies like https://www.nexvortex.com/ at 2.5c. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, May 10, 2006 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-Users] voipjet down? And again I'll say... calleveryone.com for all your RELIABLE termination needs. And again... don't go by the rates on the page... those are the end-user rates... call them for wholesale rates.. they will be competitive to voipjet, and you get phone support and quick response time. Come on guys... if you are still using VoipJet, you don't care about your companies termination. On 5/9/06, Wes Baehr [EMAIL PROTECTED] wrote: Even stranger is when calls (to the same server) work from one asterisk server account, but fail from another asterisk server account. Sometimes changing the server helps, sometimes it doesn't. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 Fax: 330.882.0455 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, May 09, 2006 4:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voipjet down? I havebent been able to call out in weeks and nobody returns emails to [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com Sent: Tuesday, May 09, 2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voipjet down? Somebody know if they are down? Let me know, Julius C. Barber [EMAIL PROTECTED] www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 5/8/2006 ___ --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 ___ --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
RE: [Asterisk-Users] voipjet down?
You always recommend CEO when ever anyone asks about a service provider when nobody has asked about wholesale. Very very few people are interested in wholesale pricing. If you are a consultant you want to recommend a company to your clients that is the best fit for their needs. If you are an end user you also do not care about wholesale pricing. So yes, everytime someone asks for a recommendation you throw this incredibly expensive provider out there and if anyone points out the fact that they are the single most expensive provider available you turn around and say but that's not their wholesale price. Well, it IS the price they advertise on their website, and in fact, it is the ONLY price advertised on their site. So you just keep extolling the virtues of them and I will point out that it is not a good recommendation if they are looking for a cost effective voip service provider. Fair enough? Secondly, read your own message, you recommend voipjet for both situations when everyone is having issues with them. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, May 10, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voipjet down? I'm not in bed with CEO. All I do is use them... and want to see them get more customers. Sorry if I sounded upset... it just seems like Kerry never reads the e-mail every time this issue comes up. On 5/10/06, Jay Milk [EMAIL PROTECTED] wrote: Whoa, kid, do you have a behavioral problem? No reason to get nasty. You're only confirming the long-held suspicion that you're in bed with calleveryone. Friendly references are one thing, but defending it at the expense of courtesy, that's suspicious. I don't think their customer service is all that top-notch anyway. I emailed them a pre-sales question on a weekday, and didn't get a response until almost 72 hours later. Maybe their customer service is better than their sales team, but in my experience, sales are generally the best staffed group in any service company. Lastly, the customer service quality of my termination provider is secondary. A business that relies on outgoing calls wouldn't even touch IP termination for those essential calls; they'd get a PRI. Everyone else can wait a few minutes while the phone system switches providers if one goes down. Matt wrote: Kerry, Do you have a reading problem? Both times that I have tried to help people out by suggesting a company I have personally used and have had good luck with, you reply and say that the rates are horrible. If you would read my e-mails you would see that the 3.9 cents is NOT for wholesale termination. If you want someone would will give cheap termination to end users, go use voipjet or whatever you want. If, on the other hand, you want some reliable cheap wholesale termination, go check out voipjet. On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote: I cant imagine anyone using voipjet as their only or main provider. And I'll say again, 3.9 cents for an ITSP is the most expensive I have found. Business grade termination is typically much less than that with top notch companies like https://www.nexvortex.com/ at 2.5c. ___ --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 ___ --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] Web Admin
You could install any number of interfaces but it does not come with one. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francisco SalinasSent: Wednesday, May 10, 2006 10:05 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Web Admin I am planning to deploy Asterisk Business edition. Does this edition have a web module administration? Thanks ___ --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] ethernet interface shares interrupts with tdm card
Go into the BIOS and disable every possible device like USB, COM, Serial, etc. But odds are, you are screwed with that motherboard. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonio AlmodóvarSent: Wednesday, May 10, 2006 8:01 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] ethernet interface shares interrupts with tdm card Hello.I have a MinITX motherboard with only one pci slot and one onboard ethernet interface, I have a TDM04B card plugged into that motherboard and the /proc/interrupts:CPU0 0: 169626332 XT-PIC timer 1: 1270 XT-PIC i8042 2: 0 XT-PIC cascade 8: 4 XT-PIC rtc12: 170166219 XT-PIC eth0, wctdm14: 398500 XT-PIC ide0 NMI: 0 ERR: 0I've tried modifying parameters in the bios and I didn't managed to change the irq.Does anyone have a machine like mine?Have anyone changed the irq in order to not sharing irq's? Thanks in advance. ___ --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] VOIP provider
Have you looked at CBeyond? I like their T1 SIPConnect product. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] VOIP provider Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin ___ --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 management interface
[EMAIL PROTECTED] is an ISO image that installed CentOS, Asterisk, FreePBX, and some other tools. FreePBX is just the web interface. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zach A Sent: Tuesday, May 09, 2006 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk management interface How different is FreePBX from Asterisk @ Home? Zach ___ --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
RE: [Asterisk-Users] voipjet down?
I havebent been able to call out in weeks and nobody returns emails to [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.comSent: Tuesday, May 09, 2006 12:40 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] voipjet down? Somebody know if they are down? Let me know, Julius C. Barber[EMAIL PROTECTED]www.GringoTel.comTel. USA: 1-408-705-1189GringoTel - ahorre en sus llamadas internacionales. ___ --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 management interface
Why make a brand new? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Sunday, May 07, 2006 11:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk management interface Hi, I have to make a web-based management interface of configuring asterisk i wanted to know if it is as simple as reading the .conf files and searching for the required section in the file and adding users etc. or there are other steps involved too?? As I have seen many such built codes on this site and found lots of code... kindly tell me how complex it is and how many other steps are involved in making this interface as i am new in this. Emmo. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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
RE: [Asterisk-Users] asterisk management interface
http://www.freepbx.org Why would you need to create your own? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Monday, May 08, 2006 12:01 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] asterisk management interface As I know only php and no other langugae like perl or any other... most of the links to such applications i have seen on voip.org site made in php are removed or are inactive. Can you tell me of any such application that i can use or make my own using that made only in php and serving my pupose? thanx! From: Kerry Garrison [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] asterisk management interface Date: Sun, 7 May 2006 23:45:58 -0700 Why make a brand new? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Sunday, May 07, 2006 11:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk management interface Hi, I have to make a web-based management interface of configuring asterisk i wanted to know if it is as simple as reading the .conf files and searching for the required section in the file and adding users etc. or there are other steps involved too?? As I have seen many such built codes on this site and found lots of code... kindly tell me how complex it is and how many other steps are involved in making this interface as i am new in this. Emmo. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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
RE: [Asterisk-Users] asterisk management interface
Those are reasons for WANTING to create your own, he specifically said I HAVE to make my own and I wanted to know why he HAS TO create his own when there are fantastics products already available. There is a huge difference in saying I would like to create my own and I have to create my own. I totally understand the 'want', I want something that is different and don't the way I want but I don't need to right now. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Monday, May 08, 2006 2:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] asterisk management interface [EMAIL PROTECTED] wrote: http://www.freepbx.org Why would you need to create your own? Many reasons: 1. not relying on already busy open source developers 2. creating something that you can possibly offer as your own commercial offering 3. have it designed exactly they way you want it from ground up 4. have a lot fun with it (and headaches :) ) etc... It is a long road though. We started PBXware in 2003 and there are still many features we wish to implement. Senad ___ --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
RE: [Asterisk-Users] asterisk hardware
He asked about hard phones not soft phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 07, 2006 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk hardware Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Tofik Suleymanov wrote: Hello folks, anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. Thank you, Tofik Suleymanov ___ --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
RE: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0
Go into the BIOS, disable every possible device such as floppy controller, usb, serial, parallel, etc. If that doesn't work, move card to another slot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Friday, May 05, 2006 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0 Hi Phil, may sound a stupid advice buthave you tried to change PCI slots on your server? Giorgio Phil Menico wrote: I have a conflict problem with the eth0 card and wct2xxp digium board. The PRI can receive calls but my network connection is gone. When I cat /proc/interrupts I get the following: 1 .. 1 .. .. .. .. 169 0 IO-APIC-level wct2xxp, eth0 .. etc. even before I modprobe wct2xxp After I modprobe wct2xxp and modprobe wctdm and again run cat /proc/interrupts I then get: .. .. .. .. .. 169 118489 IO-APIC-level wct2xxp, eth0 201 118497 IO-APIC-level wctdm .. etc How can I force the wct2xxp to load on a separate IRQ? I tried moving the eth0 to IRQ 10 but could not. Any ideas? Thank you. _*/Phil Menico/*_ XTEND Communications 171 Madison Avenue, New York, NY 10016 212-951-7632 (Office) 212-951-7683 (Fax) www.xtend.com -- -- ___ --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 ___ --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] CW options not changing
I have a weird issue with a system, running freepbx in devices and users mode (same as dozens of systems) but this one when you hit *70 and get "call waiting activated" it is not storing the setting in the database. I can manually set CW 900 ENABLED but then *71 does not disable it. Any suggestions? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Voipjet Problem?
Hard to believe you arent associated with calleveryone.com as I find it hard to believe that you would be extolling the virtues on one of, if not the most expensive companies around. $7 a month plus 3.9 cents a minute domestic, that's pretty much double the cost of anyone else. Customer service may be stellar but when clients are actually trying to save money, that's a damned hard sell. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, May 04, 2006 8:18 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipjet Problem? Just wanted to add my 2 cents. We were with voipjet, and do still use them for occassional backup.However, their lack of personal service and inability to get ahold of someone drove us away.After several total blackouts (like what happened yesterday), and no responce we finally put out an SOS on the asterisk mailing list. Of course there were several responces from companies trying to solicit us. but the one that caught our attention was calleveryone.com So far we have been rock-solid-happy with them. We've had a few small bumps along the road. For instance, once there was a router along our path to them that was dropping packets, but this was quickly resolved. Additionally, they've worked with us on the phone to resolve audio problems, and diagnose carrier issues. If I have a problem, I rest assured that I can call someone, or page someone if the situation is severe enough, and get ahold of a human at any hour of the evening. Not so with VoipJet. I don't want to bad mouth VoipJet, their service is decent... but definately not acceptable for a carrier grade level. I'm not affiiliated with calleveryone in any way other then a very happy and satisfied customer, and would highly recommend them to you. If you are a wholesole buyer of minutes, talk to them, don't just take their prices on the main page... those are for residential and regular customers. Their prices are very comparable to voipjet, and the service is miles ahead. On 5/3/06, Matt [EMAIL PROTECTED] wrote: Yup... I think they died... this is why I stopped using them except as my backup. It seems 64.34.45.100 is working ok as of right now. It wouldn't be so bad if they had a number you could call for support! HERE THAT JOHN? You need a phone number if you want to play with the big dogs. On 5/3/06, Mark Hulber [EMAIL PROTECTED] wrote: I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers. MARK. ___ --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 ___ --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] Simple Dell Computers
Most of the Dells will work fine with some minor workarounds. First off, go into the BIOS and disable every possible device (USB, Floppy controller serial, parallel, etc). Then if the card does not work correctly, move it to a different slot. With most of the lower end Dells you will find that the card will only function properly in one of the three PCI slots. If you get a motherboard that has more than 3 PCI slots your chances of success are dramatically higher. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Wednesday, May 03, 2006 5:10 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Simple Dell Computers Hello List, I know this has been brought up many times but I wanted to know if anyone had any expirience in the following. I setting up several voice mail systems. Each one is going to have a TDM400P. Two FXO for people to leave messages and two FXS for POTS phones so people can listen. Anyone know if there are any simple specific dell models that will handle this without a problem ? Thanks. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
RE: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset
Are you seriously trying to run 4 cards in one system? The odds of getting that working are about the odds of Angelina Jolie showing up on my doorstep ready to whisk me off tobut I digress...you will have serious interrupt issues trying to get 4 cardss working in one system. I am surprised that you would fork for a PRI card but use cheap winmodems for analog lines. You will have much better luck tossing the x100p cards and using either SPA-3000's, a TDM400, or a Mediatrix 1204. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Tuesday, May 02, 2006 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? ___ --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
RE: [Asterisk-Users] FreePBX in production?
You can already do that. You ca specify different access to different users with the Administrators module. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, May 01, 2006 6:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX in production? Time Bandit wrote: You could, but it'll get overwritten by any FreePBX upgrades. The *.conf and *_additional.conf files are controlled by FreePBX and can be overwritten. I thought I should clarify this statement: I meant that FreePBX could overwrite both the *.conf and the *_additional.conf files. You are strongly advised NOT to edit either of those types of files. All editing should be restricted to the *_custom.conf files. Well, I've modified *.conf files and I never had AMP (FreePBX) overwrite them. An upgrade would most certainly overwrite them but not normal usage. I may be wrong, but if you upgrade, the *_custom.conf files will probably get overwritten also, so you better backup them before. Let's see if I can summarize various recent postings relative to the broader topic of whether FreePBX/AAH is production-ready. Seems the general consensus is that AAH and/or FreePBX is considered production ready if the functionality embedded in AMP (primarily) happens to fit the specific small business requirements. Anything outside of the basic functionality is limited primarily by the lack of technical documentation, the undocumented logic behind magically creating dialplan entries, and limitations associated with AMP interfaces to various channels such as those typically defined in zapata.conf, etc. It would almost appear as though the user interface should be broken into two components: 1) a simplified interface for non-technical users that are responsible for adds/moves/changes, and, 2) a second interface to define business-specific items such as defining certain interfaces (eg, zap channels), contexts, dialplans, etc. Many of those items defined in #2 would probably become drop-down selections for the user interface in #1. Thoughts? R. ___ --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
RE: [Asterisk-Users] Softphone ready to go installed on USB flashdrive
The current versions of IDEFISK use a Windows installer, wether it is required or not now I dont know. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Softphone ready to go installed on USB flashdrive I do this with the windows version of idefisk from Asteriskguru.com. The configuration is stored in the dir with the program and dll. I have actually configured it and emailed it to users. There is no installer and a simple shortcut or autoplay menu should take care of the rest. On 5/1/06, Time Bandit [EMAIL PROTECTED] wrote: How can I install a softphone on my USB flash drive like Xlite and have it ready to go when I plugit in at any Windows XP computer? (Same for a Linux softphone, both on one USB flash drive).I believe Dan's softphone is suitable for this. See http://www.laser.com/dante/diax/diax.htmlActually, I should do that with my softphone instead of using the registry :(hth___ --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 ___ --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 in production?
There are still some basic things missing (for example if you don't use voicemail it is not possible to set a destination for the call if not answered, you have to create a ring group for each extension to work around it, this is a major issue) Remco - take a look at the Follow Me module I added. It is basically a presonal ring group for each extension. If you want to do the above, just define the Follow-Me settings to ring your own extension (or more if you want) and then choose any destination you want. It effectively does 'creat a ring group for each extensions' that wants one, but it does it in such a way as to be separate and work side by side with normal ringgroups, and there is a direct link between it and the extension (or user) so that navigation is very easy as you can bounce back and forth with a single mouse click. Many people have talked about limitations of freePBX and how you cant do custom things. Both Phillip and I attacked the follow-me function this week with his using personal ring groups and mine using personal call queues (see article at http://voipspeak.net) to simulate the functionaly of the locate function from CallManager. Both solutions used only freePBX functionality to acocmplish two relativly complex tasks that many people have been struggling to create with just the config files. That speaks to the flexibility of using the system. Sure there are bound to be limitations that may prevent some type of functionality, no system is perfect, but it does cover far more than just a few small businesses. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --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] RE: Asterisk is stripping my area code
Are you dialing 9 first? It is showing that the digits you
dialed are:
9-770-719-0239
Using your dialplan you should be dialing
1-770-719-0239
Kerry
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim
LynchSent: Sunday, April 30, 2006 10:10 AMTo:
Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE:
Asterisk is stripping my area code
I don't know if this helps, from the log.Jim.Apr 30
12:58:55 VERBOSE[4225] logger.c: -- Executing
Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new
stack
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30
12:58:55 DEBUG[4225] chan_zap.c: Deferring
dialing...
Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called
1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19,
channel
1
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index
0)
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RE: [Asterisk-Users] RE: Asterisk is stripping my area code
Do you have 9 as a prefix in the trunk? It is actually
ADDING a 9 to the phone number before it dials.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim
LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] RE: Asterisk is stripping my area code
No, I'm just dialing 7707190239. When I tried it with a 1, it
gave me the same result, a nice lady telling me "when making a local call you
must first dial the areacode" or words to that effect. From
the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c:
Function result is '"" 7707190069'Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing NoOp("SIP/200-fa0b", "CallerID
set to "" 7707190069") in new stack Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing Set("SIP/200-fa0b",
"GROUP()=OUT_1") in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function
result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is
'0'Apr 30 13:29:04 VERBOSE[4242] logger.c: --
Executing GotoIf("SIP/200-fa0b", "0?108") in new stackApr 30 13:29:04
DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing Set("SIP/200-fa0b",
"DIAL_NUMBER=17707190239") in new stackApr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing Set("SIP/200-fa0b",
"DIAL_TRUNK=1") in new stack Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing AGI("SIP/200-fa0b",
"fixlocalprefix") in new stackApr 30 13:29:04 VERBOSE[4242]
logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242]
logger.c: fixlocalprefix: Removed prefix. New number:
7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c:
-- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04
VERBOSE[4242] logger.c: -- Executing
Set("SIP/200-fa0b", "OUTNUM=97707190239") in new stackApr 30 13:29:04
DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing Set("SIP/200-fa0b",
"custom=ZAP/1") in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_
result is '0'Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Executing GotoIf("SIP/200-fa0b", "0?16")
in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any
branchApr 30 13:29:04 VERBOSE[4242] logger.c: --
Executing Dial("SIP/200-fa0b", "ZAP/1/97707190239|120|W") in new stackApr
30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04
DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242]
logger.c: -- Called 1/97707190239Apr 30 13:29:05
DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05
DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1
(index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel
1So what should I have in my dialing plan to let me dial 7707190239
and have it use that exact number? Or do I have to dial 9
first?Thanks, Jim.
On 4/30/06, Kerry
Garrison [EMAIL PROTECTED]
wrote:
Are you
dialing 9 first? It is showing that the digits you dialed
are:
9-770-719-0239
Using your dialplan you
should be dialing 1-770-719-0239
Kerry
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200-
[EMAIL PROTECTED]
http://www.techdatapros.com
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of
Jim LynchSent: Sunday, April 30, 2006 10:10
AMTo: Asterisk-Users@lists.digium.comSubject:
[Asterisk-Users] RE: Asterisk is stripping my area
code
I don't know if this helps, from the log.Jim.Apr
30 12:58:55 VERBOSE[4225] logger.c: -- Executing
Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new
stack
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30
12:58:55 DEBUG[4225] chan_zap.c: Deferring
dialing...
Apr 30 12:58:55 VERBOSE[4225] logger.c: --
Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on
19, channel
1
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index
0)
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RE: [Asterisk-Users] FreePBX in production?
Its just you. There is much more flexibility on handling incoming pstn lines than there was in the last version of AMP If you like manually creating config files with custom settings for each user, then a GUI is not for you. I have several clients using freePBX because it is easier to maintain some of the features they wanted this way than dealing with the config files. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, April 30, 2006 2:20 PM To: Asterisk Users-List 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
RE: [Asterisk-Users] queues
This is not the right place for help with AAH. Use the AAH forum at sf.net. If it is just hanging up on users, it is not configured properly. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Siglin Sent: Sunday, April 30, 2006 8:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queues I am not understanding how queues are supposed to work. I am using [EMAIL PROTECTED] and configured a queue in AMP. I have also set my static extensions in the queue. If I set up the system to put people in the queue on incoming it just hangs up on them. If I try to log in as an agent it says I am logged in and then disconnects. If I do a show agents it says I'm not logged in. I looked at some samples but not quite getting it. My queue is 100 and my two extensions are 200 and 201. ___ --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
[Asterisk-Users] Locate Me Function with freePBX
The client's needs are the mother of invention. We have a client that currently uses a Cisco Call Manager and one of the features they love was the Locate-Me function (or follow-me, or find-me, whatever you want to call it) which basically rings their desk phone a few times then plays a short message, and then rings their other remote phones and cell phones. The customer wants this same functionalty from an Asterisk system that we will be building running freePBX. It took me a while to think about how to implement this without mucking around with the config files but the end result is a fairly simple solution that enables the use to turn on/off the function at will. For the complete article on how to implement this feature, go to:http://voipspeak.net/index.php?option=com_contenttask=viewid=72 Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] RE: Install/Upgrade
upgrading from what version? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Saturday, April 29, 2006 6:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] RE: Install/Upgrade Hi all, I was just wondering ifanyone knows of any gotchas with respect to upgrading Asterisk to the latest 1.2.7 ? Is the procedure the same? Config files remain intact? Just untar/make install? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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 use a softphone -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Thursday, April 27, 2006 9:11 PM To: Steve Feinstein Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] USB conference phone OK, assuming the usbaudio sees the conference phone and can work it, how would you write an extension to ring that? on Thursday 04/27/2006 Steve Feinstein([EMAIL PROTECTED]) wrote It's a standard USB audio device. While I haven't tried it, I'm pretty sure the Linux USB audio driver will probably see it. -Steve John covici wrote: Any way to use this on a Linux box so I could use this with asterisk? I have a windows box on the same network, but how would I get asteriskto see such a thing? Thanks. on Wednesday 04/26/2006 Steve Feinstein([EMAIL PROTECTED]) wrote http://www.iogear.com/main.php?loc=productItem=GPH100U I've got a couple of these, they're $40 and there's a $20 rebate going on now. For that price it's pretty amazing. Plug and play, no drivers required. Quality very good, it does echo cancellation and noise reduction. I only wish it had a mute button. BTW: I don't have any affiliation with ioGear other than I like this product. Jim Houser wrote: 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 a hardphone 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 Collins *Sent:* Wednesday, April 26, 2006 10:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* 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 Garrison *Sent:* Wednesday, 26 April 2006 10:36 AM *To:* '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 http://voipspeak.net/index.php?option=com_contenttask=viewi d=39Itemid=27 -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jim Houser *Sent:* Wednesday, April 26, 2006 6:26 AM *To:* '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 http://www.provantage.com/usb-internet-phone%7E220150620.htm It operates nice and has very good call quality. -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean Collins *Sent:* Tuesday, April 25, 2006 8:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset- free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDV WQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid =39Itemid=27 http://voipspeak.net/index.php?option=com_contenttask=viewi d=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
RE: [Asterisk-Users] FW: NuFone Update: DIDs
AMEN!! Any consultant that DOESNT take this into consideration should stick to installing Windows and calling themselves an IT "Expert". You can screw up someone's network, mess upa workstation, hose their email, but you break someone's telephone service there will be hell to pay. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - AspendoraSent: Friday, April 28, 2006 4:39 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] FW: NuFone Update: DIDs Exactly why I chose to go with a PRI for business use. There is something to be said for the stability of a telco, even if it's not SBC (or ATT now). In some cases, government interference is good. How many businesses can survive a loss of their phone number? I know the ones I deal with cannot. This is something we need to take into consideration as Asterisk users or consultants. We need to look at the whole picture, not just a short term savings. Can your business/clients survive without listings in directory assistance or the phone books? Can they survive if they have to change numbers due to their Voip provider losing a contract? Some can, most can't. I looked into using Voip. Technically, it seemed like a good solution. I just don't trust it in the long run. I know that by using a telco, I will have access to my phone numbers. With Voip providers, who controls the numbers? I think this is something a lot of people fail to take into consideration. On 4/28/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I would be very wary, as VOIP providers feel no responsibility to thecustomer and will not bother to tell us they might not be around next week. Once bitten...--Chris MasonNetConcepts(264) 497-5670 Fax: (264) 497-8463Int:(305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --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] Polycom NTP issue
I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Polycom NTP issue
Polycom 501 Firmware: 1.6.2.0041 Bootrom: 3.1.0.0269 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Thursday, April 27, 2006 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom NTP issue What Polycom phone model? What firmware version? What bootROM version? Older versions of Polycom phones only worked with SNTP time servers not NTP. MATT--- On 4/27/06, Kerry Garrison [EMAIL PROTECTED] wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --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 ___ --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] Polycom NTP issue
DNS is Windows 2003 Using the NTP server from CentOS 4.3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Thursday, April 27, 2006 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom NTP issue What dns server are you running? On Thu, 27 Apr 2006, Kerry Garrison wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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
RE: [Asterisk-Users] Polycom NTP issue
Here is the sip.cfg file SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=192.168.10.50 tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ Using the same IP for the server on a Linksys SPA-941 everything is correct, using this configuration on the 501 shows 24 hours ahead. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, April 27, 2006 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom NTP issue Kerry Garrison wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? This is a copy/paste of the exact statements used on a IP600 phone: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=134.84.84.84 tcpIpApp.sntp.gmtOffset=-21600 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Are you sure you are setting the gmtOffset to the proper number? The above example is for CST, which is -6 hours (or -21600 seconds) from GMT. It is also config'ed with resyncPeriod = 24 hours, meaning the clock is only sync'ed once per day. What are you using for the above? ___ --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
RE: [Asterisk-Users] Looking for input on which way to go with smallbusiness setup
You will kick yourself up and down the block for not using IP phones in the end. What are you going to spend? $65 for an ata and $25 for a phone? Spend the extra money and get SNOM or Linksys phones. You then need to figure out how to get 12 analog lines into Asterisk. Using 3 TDM400's is not really an option as you will spend countless hours trying to figure out interrupt issues. The next best option is 3 Mediatrix 1204 Gateways, this will set you back about $1,800 and will be maxed out. The best option, sticking with analog lines that is, would be a Rhino CB-24 FXO Channel Bank connected to a Rhino R1T1 card in the asterisk server. This bundle will hit you for about $2,000 but will only be half full. However, you say you have 12 extensions and 12 lines? It is very rare to have a 1:1 ratio. Usually for 12 people you will see 5-8 lines needed. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: Thursday, April 27, 2006 1:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for input on which way to go with smallbusiness setup Hey guys! I'm the past week and a half, I have really learned a lot from the mailing list and the wiki's posted online. Now I have a question regarding different ways I can setup my asterisk server for a small business with 12 extensions in the office. Cost is a great concern, so I know cheap analog phones at the desks is what we are looking at. My question is, should I go do a fractional T1 for voice only, and the get a Internet Service provider for their data needs, get some sort of ATA and run the 12 analog phones to that OR go with what SBC/ATT (in my area) is offering, and do the Integrated T1 option and have just the 12 channels for voice, run some ATA between * and the phones and use the 768 up and down for data? Anyone use this Integrated T1 with asterisk before? What hardware did you use? Thanks for your input! Terrelle ___ --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
RE: [Asterisk-Users] Polycom NTP issue
I changed the DHCP overrides to 1, restarted the phone, now it showing the right day but it is 4 hours and 29 minutes fast. Arrrgh. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, April 27, 2006 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom NTP issue I don't have a polycomm manual handy, but I think I'd change the overrideDHCP parameter to 1 and test. You are apparently in the PST timezone? If that doesn't do it, my next step would be to use ethereal to capture one of the ntp request/response pkts and analyze the content. If that looks okay, then something in the phone isn't right. Kerry Garrison wrote: Here is the sip.cfg file SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=192.168.10.50 tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ Using the same IP for the server on a Linksys SPA-941 everything is correct, using this configuration on the 501 shows 24 hours ahead. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, April 27, 2006 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom NTP issue Kerry Garrison wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? This is a copy/paste of the exact statements used on a IP600 phone: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=134.84.84.84 tcpIpApp.sntp.gmtOffset=-21600 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Are you sure you are setting the gmtOffset to the proper number? The above example is for CST, which is -6 hours (or -21600 seconds) from GMT. It is also config'ed with resyncPeriod = 24 hours, meaning the clock is only sync'ed once per day. What are you using for the above? ___ --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 ___ --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
RE: [Asterisk-Users] Looking for input on which way to gowith smallbusiness setup
What would the difference be with using IP phones or ATA's in that case? You are still talking about a network device. Even with 50-60 stations installs you are highly unlikely to run into a need for QoS internally. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Thursday, April 27, 2006 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for input on which way to gowith smallbusiness setup Kerry Garrison a écrit : You will kick yourself up and down the block for not using IP phones in the end. What are you going to spend? $65 for an ata and $25 for a phone? Spend the extra money and get SNOM or Linksys phones. You then need to figure out how to get 12 analog lines into Asterisk. Using 3 TDM400's is not really an option as you will spend countless hours trying to figure out interrupt issues. The next best option is 3 Mediatrix 1204 Gateways, this will set you back about $1,800 and will be maxed out. The best option, sticking with analog lines that is, would be a Rhino CB-24 FXO Channel Bank connected to a Rhino R1T1 card in the asterisk server. This bundle will hit you for about $2,000 but will only be half full. True, but using IP Phones means you need to do proper QoS at the level of your LAN or install a second LAN (which kinda defeats the purpose) as otherwise a network card going bananas could kill your phone system. So you need to get a proper QoS enabled switch, which is quite expensive too... ___ --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
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. ___ --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
Yes, I have that device, I wrote the review of it and have used it regularly ever since. I use it with IDEFISK softphone for the most part but have tested it with Skype, X-Lite, and SJPhone. I have had it since November and just love it. Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Wednesday, April 26, 2006 8:24 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. ___ --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
it was a revewiers sample that I begged them to not make me send it back and they let me keep it. Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Wednesday, April 26, 2006 4:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] USB conference phone Lol now the important question.Did you pay for it or was it a reviewers sample J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry GarrisonSent: Wednesday, 26 April 2006 7:23 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB conference phone Yes, I have that device, I wrote the review of it and have used it regularly ever since. I use it with IDEFISK softphone for the most part but have tested it with Skype, X-Lite, and SJPhone. I have had it since November and just love it. Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Wednesday, April 26, 2006 8:24 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. ___ --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] Paging on Aastra analog phones.
When used in this mode they can only detect a ring. Your best bet would be to put in some overhead paging. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard SchroederSent: Wednesday, April 26, 2006 6:51 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Paging on Aastra analog phones. Hello All, We have an installation that has Aastra analog phones connected to the asterisk server with Sipura ATA devices. (It was done this way in order to use existing wiring). Is there any way to implement a page all by turning on the speakers in these phones (like you can do by turning on auto-answer on some SIP phones)? Thanks in advance for your help. R C Schroeder ___ --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 as a phone survey system
Asterisk is simply a telephony toolkit, so the simple answer is yes, Asterisk can do this. Also, being a toolkit means there are a number of ways to accomplish it. You could right PERL, Python, TCL, C, PHP or numerous other types of scripts that can manage this for you. To see how to do some of the basic functions, you can look at some of the scripts at Nerd Vittles (http://nerdvittles.com). Things like the TeleYapper will give you a basis to work from. Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TV JOESent: Wednesday, April 26, 2006 7:31 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk as a phone survey system Hi,I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on thetouch tone response another message would be played. Is it possible for asterisk to manage a survey like this? If so can the responses from the listeners be recorded. If someone else has done this I'd be interestedin details.TIA , TV JOE Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates. ___ --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] About Softphone IAX free for Pocket PC
Unless you have a top of the line Pocket PC don't even bother. Most inexpensive units like the T-Mobile MDA just dont have the processing power to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I forgot about already and the sound quality was horrible regardless of using GPRS or WiFi. That would have been a great benefit to me but its just not going to happen on a device that barely runs Windows Mobile as it is. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy Sent: Tuesday, April 25, 2006 8:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] About Softphone IAX free for Pocket PC Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmerón Ntutumu[EMAIL PROTECTED] Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de países en desarrollo ___ --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
[Asterisk-Users] Auto Logout from queue
i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Splitting Zap channels into trunks?
On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Auto Logout from queue
Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Tuesday, April 25, 2006 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Auto Logout from queue Use the local channel to call the agent first, and if there is no answer, log them out. _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com attachment: winmail.dat___ --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] Polycom MWI
Thanks a ton!! When using Extensions mode (the default) this would be: [EMAIL PROTECTED] When Using Users and Devices mode this would be: [EMAIL PROTECTED] Thanks for the guidance there, this has been driving me nuts. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, April 21, 2006 5:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom MWI Ohh yeah good point. I had a similar issue when I started using FreePBX and it didn't fill out the mailbox field automatically. Once I added the [EMAIL PROTECTED] there the MWI started working as well. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Friday, April 21, 2006 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom MWI Kerry Garrison wrote: Didn't help. Could I be missing something else? My phone.cfg looks like this: mwi msg.mwi.1.subscribe=300 msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97/ And sip.conf for extension 300: [300] username=300 type=friend secret=*** record_out=Adhoc record_in=Adhoc qualify=no port=5060 pickupgroup=1 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup=1 callerid=Polycom IP501 300 allow=alaw allow=g729 Mine works fine, so I hope that helps. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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 ___ --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] Announcement System for a Charity
NerdVittles.com has a dialout announcement system article. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Thursday, April 20, 2006 11:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Announcement System for a Charity I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to modify their announcements. Before I write it, I was wondering if anyone had an extensive dialplan or an AGI script that already did something like this. I know it'll only take a couple of hours to write and test this, but I thought if someone has something already written, I could just borrow it from you. Thanks, Nabeel ___ --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
[Asterisk-Users] Polycom MWI
I have tried everything from voip-info and I still cant get the Polycom 501/601 to display the MWI indicator light. Everything else works just fine. I am using FreePBX set to users and devices mode. Here is the MWI section of the phonexxx.cfg file: mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*97" msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="disabled" msg.mwi.3.callBack="" msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="disabled" msg.mwi.4.callBack="" msg.mwi.5.subscribe="" msg.mwi.5.callBackMode="disabled" msg.mwi.5.callBack="" msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/ /msg i have also tried msg.mwi.1.callBackMode="register" Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Polycom MWI
Didn't help. Could I be missing something else? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, April 20, 2006 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom MWI Put your voicemailbox number (usually extension) in the 1.subscribe field. Bill _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Thu 4/20/2006 7:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom MWI I have tried everything from voip-info and I still cant get the Polycom 501/601 to display the MWI indicator light. Everything else works just fine. I am using FreePBX set to users and devices mode. Here is the MWI section of the phonexxx.cfg file: mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97 msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg i have also tried msg.mwi.1.callBackMode=register Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com attachment: winmail.dat___ --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] need stand-alone FXO ports
Linksys SPA-3000 Single Port $90 Mediatrix 1204 4 port Gateway $580 Rhino CB24 24 Port Channel Bank + Rhino R1T1 Card $2000 Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 19, 2006 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] need stand-alone FXO ports What are you using as FXO ports for a few analog (remote) lines? What is the price, where to buy, what is your experience? 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 ___ --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 no sound from sound card
I don't know that distro but with CentOS you have to run alsamixer to turn on the output and turn up the volume, it is off by default. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Hodder Sent: Thursday, April 13, 2006 9:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk no sound from sound card Hi all, I have a k6 2-400 that has a creative awe64 sound card. It plays sound fine when using a standard audio player. I want to use the sound card as a console phone but there is no audio from the sound card. I have tried compiling various version of asterisk with no luck. I did use this machine about 2 years ago with Asterisk and Mdk 10 and it worked fine. Currently using Mandriva 2006. Any ideas in getting the sound card to work would be apreciated. Thanks Gary. ___ --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
RE: [Asterisk-Users] freepbx dialing prefix
Submit a bug report to the FreePBX team? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean GarlandSent: Wednesday, April 12, 2006 8:46 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] freepbx dialing prefix I need to put a w in the dialing prefix, but it says it isnt valid. If I manually modify the extension file, it then affects all calls made over any trunk. Any ideas? Sean --No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 ___ --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
[Asterisk-Users] SPA-941/942 Bulk provisioning
Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Force codec
Disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael StrelnikovSent: Saturday, April 08, 2006 7:25 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw.Thanks.Best regards,Michael ___ --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] Instant Message?
I tried the latest version of Jive over the weekend and I have to say it is a giant pile of crap. I did this on multiple machines on both Linux and Windows, and after setting everything up, the moment you add the asterisk module, all authentication and user setup is lost and there is no way to log back in as the admin to fix it. If anyone has any more positive experience I would like to hear about it as it sounds very interesting. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, April 09, 2006 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Instant Message? Zhiqiang Li wrote: Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any related documents or weblinks? -- Thanks Best Regards! Steven Li I am not sure exactly what you are trying to do but Jive Messenger has asterisk add-ons and functionality. Might be worth a look for ya. Thanks, Steve Totaro ___ --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
RE: [Asterisk-Users] Using Call Progress
Welcome to the painful world of analog phone lines. Unless you are using a digital line, there really is no true call progress detection available. In many situations this is not a problem, where we see this the most is when you are trying to ring a zip device and a zap channel at the same time, the zap call progress indicates an answered line the moment the zap channel goes active, NOT when the far side answers. If you have a ring group with sip and zap channels, what typically happens is that the sip phone will ring once, but as soon as the TDM card places the outbound call, it is considered "answered" and the sip phone stops ringing. Yes, you can enable callprogress and several other tweaks but the end result is often the far side answering and Asterisk still playing ring tones because there is no signal on the PSTN to indicate a far side answer. So, what to do when you find yourself in this situation and adding a PRI is not a solution, the only way we have worked around this is to make those outbound calls over a SIP or IAX service provider (and no, using a SIP gateway like a Mediatrix 1204 does not solve the problem as it is a PSTN issue) I know some people will argue this, but this was the result of almost 12 hours of work with us and Digium to figure out this issue. After MUCH debate and many hours of testing, this became the official word. Don't shoot the messenger. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric BuruschkinSent: Thursday, April 06, 2006 6:19 AMTo: Asterisk-UsersSubject: [Asterisk-Users] Using Call Progress I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with this issue? I am using asterisk 1.2.4 with TDM04B (FXO) cards on a RHEL3 system. Something in indications.conf or zonedata.c/dsp.c in the sourcethat can be tweaked? Any help would be appreciated! Thanks! - Eric Buruschkin ___ --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 behind NAT
Yes. In Sip.conf you need the following lines: externip=xxx.xxx.xxx.xxx ; put public ip address here localnet=192.168.10.0/255.255.255.0 ; edit as appropriate In your firewall, add the following mappings to your server: 5060-5061 UDP 10,000 - 20,000 UDP Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Thursday, April 06, 2006 8:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk behind NAT Hello to all Can we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk must have a private (NATed) IP... but the idea is to make him dial other SIP domains. Can Asterisk work behing NAT, and still route calls to the Internet? And he can still receive calls from the Internet? Thanks Joao Pereira ___ --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
RE: [Asterisk-Users] increasing volume level to console/dsp
Use an amplifier off the headphone jack. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, April 06, 2006 8:30 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] increasing volume level to console/dsp I am trying to get higher volume on a console/dsp port. I have a SIP phone connected to server A, Server A has an IAX2 connection to Box B that connects me to the console/dsp port. The mixer settings are set to 90% but the audio out is not that high. How can I increase the sound level. Jerry ___ --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
RE: [Asterisk-Users] Beginner: PBX for my house
Currently Asterisk will not integrate with Skype. You would need a provider such as Teliax, Broadvoice, IAX.cc or many others or you can use hardware devices to connect to traditional phone lines. You didn't say what broadband phone you have but if its Vonage, there are also issues with "true" integration there as well. There are lots of good articles at VOIPSpeak.net and NerdVittles.com Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of erikSent: Monday, April 03, 2006 7:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Beginner: PBX for my house Hi, I am renovating my house completely and installing new cabling for communication. I'm not to into this PBX thing but I would like to have a simple one for my house, to have different phone numbers for my family members, some kind of integration with my Broadband telephone, and possibly Skype Browsing the digium website didn't make me much wiser. I have a linux box that I can dedicate to this. If I buy for example the Wildcard TE110P card, what can I as a simple homeuser do with this card? I have reasonable knowledge about networking and Linux administration, butnext to littleabout digital telephony and such. erik ___ --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] Beginner: PBX for my house
"or you can use hardware devices to connect to traditional phone lines" That can be a Digium card, Sangoma card, Linsys SPA3000, Mediatrix 1204, and several other devices. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RandyWSent: Monday, April 03, 2006 8:31 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: 'erik'Subject: Re: [Asterisk-Users] Beginner: PBX for my house What about the Asterisk Developer pack, a good Linux box and his standard phone line?? I've seen this work and it does a great job. The Telco doesn't know anything as Asterisk integrates with the analog phone line and things just work.Am I off base here??RandyWKerry Garrison wrote: Currently Asterisk will not integrate with Skype. You would need a provider such as Teliax, Broadvoice, IAX.cc or many others or you can use hardware devices to connect to traditional phone lines. You didn't say what broadband phone you have but if its Vonage, there are also issues with "true" integration there as well. There are lots of good articles at VOIPSpeak.net and NerdVittles.com Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of erikSent: Monday, April 03, 2006 7:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Beginner: PBX for my house Hi, I am renovating my house completely and installing new cabling for communication. I'm not to into this PBX thing but I would like to have a simple one for my house, to have different phone numbers for my family members, some kind of integration with my Broadband telephone, and possibly Skype Browsing the digium website didn't make me much wiser. I have a linux box that I can dedicate to this. If I buy for example the Wildcard TE110P card, what can I as a simple homeuser do with this card? I have reasonable knowledge about networking and Linux administration, butnext to littleabout digital telephony and such. erik ___ --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
RE: [Asterisk-Users] Asterisk 2.0 Where to download
Think people will fall for it again next year too? Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
RE: [Asterisk-Users] How is Teliax ?
When it works it works great. We have had a few issues lately but they were resolved fairly quickly. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Thursday, March 30, 2006 8:14 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How is Teliax ? Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk.i would like to know some feed back on "Teliax" before i purchase.suggest me if there are better sevice providers.thanksGiridhar Bandi ___ --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] AAH lost my IVR phrases
You made some change to something using AMP and it overwrote the extensions_additional.conf file as it was designed to do. The only safe place to put customizations is in extensions_custom.conf. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Hanlon Sent: Wednesday, March 29, 2006 8:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AAH lost my IVR phrases Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: Thank you for calling . I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the phpconfig for Asterisk PBX page, and selected the extensions_additional.conf page. On this page were the entries for the Normal and After Hours greetings. The initial greeting phrases were expressed in terms of statements like: exten = s,n,Background(custom/aa_num). It was easy to extend the greeting (for instance, Office hours are 7-7, Press pound for directory..) by directly adding more canned phrases, like so: exten = s, n+1, Background(custom/aa_num+1) etc... Hit update, Re-Read Configs. Try it out. It worked fine. And I felt pretty clever. For a few weeks. Then a complaint: Callers encountered an obviously truncated IVR script, and had no way out of the maze. Sure enough, only one phrase was being uttered. And, sure enough, only one phrase was being commanded by the existing extensions_additional.conf file. I re-edited the file, updated, and things worked again. !!!? What happened to my edited, updated, and Re-Read extensions_additional.conf file? Anybody ever encounter this behavior? What to do, in order to avoid this mishap in the future? Ideas, thoughts? Thanks, Jim Hanlon ___ --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
RE: [Asterisk-Users] Dumb question - reaching the PSTN
With Asterisk you can use Analog lines (PSTN) , Digital lines (PRI), or Internet Telephone Service Providers (ITSP) such as Broadvoice, Teliax, IAX.cc, and many more. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Marcus Sent: Wednesday, March 29, 2006 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dumb question - reaching the PSTN Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is more than likely on the old fashined PSTN. If I install Asterisk, how do my calls actually get completed? How do they get 'bridged' over to the PSTN? I attended a Seminar today hosted by Dynasis, and one of the issues was VoIP. ShoreTel was there, and the said I had to have phone lines, whether they were POTS lines, chennels from a T-1, whatever, we still had to have phone lines. Now I'm confused. If I implement an Asterisk based system (yes, I'd be paying a consultant to help), will I still have to maintain phone lines and pay full price for Long Distance? Simple pointers to White Papers on this issue will be sufficient. Many thanks, -- Best regards, Charles ___ --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
RE: [Asterisk-Users] Marketing Materials
http://www.asterisk.org/features -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, March 29, 2006 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Marketing Materials Darrick Hartman wrote: Bob McDowell wrote: 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. Bob, Check on Digium's website. I know there is such a creature there. Darrick Just went looking and could not find a thing. Can you give us a url? ___ --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
RE: [Asterisk-Users] Re: Best GUI for basic HostedPBX service
FreePBX allows you to set up multiple companies as well as determine what level of access each user has. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Sunday, March 26, 2006 10:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Best GUI for basic HostedPBX service In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought I'd pick a few brains first. I'm not looking to configure the Asterisk server itself, VI works adequately for that. But I want to give Web access to as many of the following features: This is something I'm will need in few months. If you find anything, please let the group know. -- Tomislav Parcina tparcina#lama.hr ___ --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
RE: [Asterisk-Users] FreePBX AAH
FreePBX is a configuration manager for Asterisk. It is NOT its own version of Asterisk, it is simply a GUI to manage the config files. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, March 27, 2006 9:53 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 ___ --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] Dell 2850 w/TDM2400?
Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] 3Com Phones
Look at the Linksys SPA942, it's a great phone for the price. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Radcliffe Sent: Sunday, March 26, 2006 10:21 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 3Com Phones Hi Daniel, If you are not locked in to an asterisk solution, I have a friend I have done a couple of network/phone systems with. I am also looking at Asterisk but have not gotten into it that far. Rich Radcliffe Kondor Waffenamt (760) 240-4728 [EMAIL PROTECTED] [EMAIL PROTECTED] 3/26/2006 9:55:38 AM Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --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 ___ --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] simple question on asterisk
Its all about how you configure your dialplan. Asterisk doesn't know what a PSTN or VOIP phone number is. If you want all 08444 numbers to go through a certain trunk, then you set your dialplan up accordingly. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hayward Sent: Monday, March 20, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] simple question on asterisk Hi, I am planning to deploy an asterisk installation but I need to convince a few managers that its a good idea. Theres something I don't quite understand though, I plan deploy a box on the end of 4 channel BRI ISDN and provide it an ADSL internet connection. Should a phone behind the asterisk PBX wish to call a VOIP phone number number, say an 0844 one from www.voip-user.org, would it send this automatically over the PSTN ISDN network or would it know to send the call over the internet. Would I need a SIP provider on the internet to forward the calls? I assume I would need some sort of directory service to know where to route the call. Thanks in advance, Mark ___ --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
RE: [Asterisk-Users] Attended call transfer with GXP-2000
If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNF and hitting Line 1 will transfer Line 2 to Line 1. Same concept as Conference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Thursday, March 16, 2006 7:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Attended call transfer with GXP-2000 Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus ___ --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
[Asterisk-Users] Asterisk Users Group Tonight, Irvine, Ca
If you are in Southern California and would like to attend the Asterisk Users Group Meeting, it is tonight from 6-9pm at the Heritage Park Library. Irvine Heritage Park Library(949) 936-404014361 Yale AveIrvine, CA 92604 Tonight we will be having a demo of SIPX, a review of the SNOM 320 phone, and a look at FreePBX, the new version of the Asterisk Management Portal. Also, more books to give away from O'Rielly!! Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Latest Dell SC430 Compatibility With Wildcard
If you go into the BIOS and disable all unneeded devices (serial, parallel, USB, floppy, etc) then you shouldn't have any problem. I have one in a 15 user setup that is working fine. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Tuesday, March 14, 2006 5:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard Anyone knows if the SC430, based on the Intel E7230 chipset, is compatible with the Digium cards? I've tried the compatibility page on digium's website. It seems like they've pulled the old compatibility list, now the links on the page only point back to the product pages. Over here, Dell is selling (for a short period of time), SC430 with Pentium D 820 Dual Core Processor 2.8GHz, 256MB RAM, 80GB SATA for about US$240. ___ --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
[Asterisk-Users] Asterisk Users Group Meeting March 16, Irvine, Ca
Irvine California, Heritage Park Library on the corner of Yale and Walnut. The Walnut is just south of the 5 fwy and Yale is between the Culver and Jeffery offramps. Meeting will run from 6 - 9pm. This week will feature a review of the SNOM 320, a demo of SIPX, some book giveaways courtesy of O'Rielly, and much more. For more information, contact me Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Analog Desktop Phone
You really aren't going to find an analog phone that works as well as a SIP phone for what you are trying to do. Some people suggested the GXP2000 for $85 which works ok in a home environment. It is not a top quality phone but it has all the features you want plus works very nicely with Asterisk. This same conversation is constantly going on on numerous forums. If you think about what you are trying to accomplish, it might put things into perspective. You are taking a state-of-the-art phone system flush with every business feature you may ever want and trying to install it into your home and you want to use a cheap phone on it. Things are just not designed that way. If you want to be happy with your system, not to mention putting some value on your time (and heaven help you if you have a wife that will use the system) you do NOT want to use a cheap phone on this system. At a minimum go with a Linksys SPA941 or a Snom 360. You will have either one working in a matter of minutes. If you don't put any value on your time, then keep monkeying around with a lesser solution, but the few hours you will save just dropping in a decent phone should more than make up for the extra cost. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thczv F. Thczv Sent: Monday, March 13, 2006 8:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Analog Desktop Phone On 3/12/06, Martin Joseph [EMAIL PROTECTED] wrote: But what the OP wanted was a sulotion that together with the SAP3000 makes for something that works even when there is a blackout, since the SPA3000 allows for failover to the FXS port from the FXO port if/when there is no power to the unit. Which makes it a very good solution when needed because of 911 reasons or the like. Actually it seems to me the Sipura 3000 is overkill in that case. There are many other ATA's that are less expensive that also have a 1 port FXS, and a PSTN failover for blackout. It seems the OP doesn't need the FXO at all? The PA168V based ATA I have does this and was a little more then half the cost of a SPA3000. Works well too. Shame on me, but I already have the SPA3000. I like it very much and it works fine. Perhaps if I need another, I will look at different products. This is for my own home, where I am keeping my POTS line, partly as a 911 solution. I have found a lot of analog desktop phones that have some of the features I want, but not all of them. The Cortelco 2200 looks like it might fit the bill. But it costs about $80. I'm not sure I want to pay that much for an analog phone that isn't wireless. Other than that, the closest I have found so far is the ATT 959: http://www.amazon.com/gp/product/B00067KETY/ref=wl_it_dp/104-2 261851-2083919?%5Fencoding=UTF8colid=1SGHZOJ18P2FBcoliid=I23 IRSR1SF2HPGv=glancen=172282 The problem with that one is that (as near as I can tell from the photos and the manual) it has no visual MWI. Still looking, Dave ___ --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
[Asterisk-Users] Echo Cancelation on TE110P
On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Multiple TDM400P's in a single machine
Yes you can but it is generally not a good idea nor is it simple to resolve the additional IRQ conflicts on some machines. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner Sent: Monday, February 20, 2006 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple TDM400P's in a single machine Am Tuesday 21 February 2006 00:24 schrieb Marc Archer: Hi All, Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P's in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration i have two tdm400p's (2xFXO, 6xFXS) in one desktop machine used as asterisk server for a small office (so the pc hardware is nothing special). This configuration is running since two weeks without any problems! Thanks, Marc. ___ --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
[Asterisk-Users] SPA-941 stutter tone
I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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 recordedcalls
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, February 14, 2006 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recordedcalls SineDialer, ViciDial, and GNUDialer can all do this. -Kerry ___ --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] AAH 2.5 pone paging broken
Using some scripts that have been posted, we have been able to get paging to phones working quite nicely. However, with a few [EMAIL PROTECTED] 2.5 installs, (Aserisk 1.2.4) the phones ring but never pick up. Any ideas on why or how to tweak the scripts to get the phone paging working again? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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 vs. Traditional PBX
Title: Asterisk vs. Traditional PBX Hi everyone !So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. [Kerry Garrison sayeth] The 841 isfine for testing but I would never put one on a clients desk. The sound quality is bottom of the barrel. Combine that with the TDM400 card and itsa wonder anyone will use the phone system at all. Move up to the Linksys SPA941 or SPA942 or the Polycom 501and then use a different interface such as the Mediatrix 1204 ora PRI and your users will be singing your praises till the end of time.So here are my questions:* Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this )[Kerry Garrison sayeth] Asterisk is a great solution for your company and you will have many more benefits than the Northstar system. * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? [Kerry Garrison sayeth] You would want a beefier machine and at least one PRI. Its not the number of people, its the number of concurrent phone calls. I see businesses with 100 people and they average 5-7 concurrent calls and I have clients with 15 people that average 12-15 concurrent calls. If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. [Kerry Garrison sayeth] My largest install is approaching 55 users, with the PRI and Polycom 501's they couldnt be happier. The system is on a nice 2.8ghz XEON system with 2gb of RAM and at peak times the server is basically idle.Thanks again this list ROCKS!Nora Lavelle ___ --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] Handset phone to replace Flash Operator Panel
The best way I have found to use FOP is on a second monitor. That way you don't need a second PC and it doesn't run behind the receptionists other applications. Just a decent LCD monitor and a second video card or a dual head video card, and you are all set. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Tuesday, February 07, 2006 9:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Handset phone to replace Flash Operator Panel Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth ___ --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
RE: [Asterisk-Users] virtual extension per user ?
This can easily be accomplished with AMP using the Users and Devices mode. http://voipspeak.net/index.php?/content/view/49/28/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Ongena Sent: Tuesday, February 07, 2006 8:55 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] virtual extension per user ? certainly on his first call, but it should be possible for him to explicitly 'register' and 'unregister' On Tuesday 07 February 2006 17:06, Joe Tahan wrote: when exactly would you like to stream this register me thingy? whenever an employee picks up the phone to dial? or when? Please specify more. Truely/ Joe From: Alex Ongena [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk asterisk-users@lists.digium.com Subject: [Asterisk-Users] virtual extension per user ? Date: Tue, 7 Feb 2006 15:26:23 +0100 Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the single extension XXX. I know you can 'forward' calls to other extensions, but when people go from Office 1 to Office 2, they forget to enable their forward in Office 1 to Office 2. I like a solution where they can say 'Please register me, I'am now sitting in Office 2'. The moment after 'registration', when you call XXX, the phone in Office 2 will ring. In all places I use Asterisk 1.2.1 with bristuff, Cisco 7940/60 phones with Sip and some Sip softphones. Any hints or tricks to get this behaviour ? Thanks Alex ___ --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 Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. -- Alex Ongena Managing Director --- Able N.V.Tel: +32(0)15 50.44.00 Dellingstraat 28bFax: +32(0)15.50.44.09 B-2800 Mechelen Belgium mailto:[EMAIL PROTECTED] http://www.axsguard.com http://www.doITsafe.net aXs GUARD - internet communication appliance --- ___ --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
