Re: [Asterisk-Users] how many potential customers out there utilizing AIX
I would think there would be good interest. Just about all the VOIP providers that have unlimited US dialing plans will not give out login information so you can't use Asterisk or your own CPE. The number of users that want to use their own CPE such as a SPA-2000 with the 2nd port configured for FWD is probably more than the number of Asterisk users right now. How long would it take for you to start offering such as service after a decision is made? - Original Message - From: dfeuer To: [EMAIL PROTECTED] Sent: Wednesday, March 17, 2004 6:38 PM Subject: [Asterisk-Users] how many potential customers out there utilizing AIX Hi Everyone, We are a service provider looking at integrating *, and notice there are a lot of issues with the company's out there that offer services with AIX. If there were a $20.00 a month program which included unified communications with the * platform or just straight termination and origination and it included all the calls in-bound and out bound in the US, and really competitive rates for outside of the US, is there a large market for these services. We would offer a one number application which would include fax with that number and a local phone number as well as offer toll-free numbers for the continental US. The service quality that we would provide has to at least be that of the local RBOC's and hopefully exceed it. It would also be easy to offer termination on a pre-paid basis say for .02 cents per min in the US through the network. Could people please tell me if it is a market worth really pursuing?? Sincerely, Don Feuer (949) 279-5290 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly Beta - with SIP and G.729
- Original Message - From: Adam Hart To: [EMAIL PROTECTED] Sent: Monday, March 15, 2004 6:32 PM Subject: [Asterisk-Users] New Firefly Beta - with SIP and G.729 Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Sounds good. Any plans for Speex codec support? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monastery devel page
- Original Message - From: Matthew Marlowe To: Asterisk Users [EMAIL PROTECTED] Sent: Monday, March 08, 2004 2:02 PM Subject: RE: [Asterisk-Users] monastery devel page I'm currently also busy to give it another look'n'feel using just some CSS. (a screenshot - all extensions blanked out - can be found at http://graphics.cs.uni-sb.de/~rainer/gui.png) We've also build a small interface to give SIP-Users the possibility to change their password. (this can be found at http://graphics.cs.uni-sb.de/VoIP/sipgui.tar.gz http://graphics.cs.uni-sb.de/VoIP/Images/gui.png) If there's interest in it I can put all the pieces together in one archive. That would be nice ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
- Original Message - From: Greg Hill To: [EMAIL PROTECTED] Sent: Wednesday, February 18, 2004 9:24 PM Subject: Re: [Asterisk-Users] Room Monitor On Wed, 18 Feb 2004, Jamin W. Collins wrote: On Tue, Feb 17, 2004 at 10:04:02PM -0800, David Liu wrote: Well use a Polycom IP 500 and put to auto answer and ringer off. Then you can use it as a room monitor device. Seems like that could do the trick. However, I was hoping for a sub $200 solution. Anyone know of a less expensive solution? If you've got an old PC lying around collecting dust (probably most of us do!) then you could install asterisk on it, configure it for auto-answer on the console, and then hook it up to your main asterisk server via sip/iax/whatever. Maybe best if it's got a quiet power supply fan.. then again, maybe the baby will appreciate the background noise. Or even easier, get a microphone and run the wire back to the sound card of your Asterisk computer. Depending on distance you may need an audio amplifier/extender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Marketing collateral, etc.
Two more you could throw in for comparison. http://www.aipcom.com/ http://www.bconvergent.net/ - Original Message - From: Ryan Courtnage To: [EMAIL PROTECTED] Sent: Tuesday, February 17, 2004 3:21 PM Subject: [Asterisk-Users] Marketing collateral, etc. Hello All, Does anyone know of a site that has gathered together marketing collateral for * ? i.e Competitive analysis, feature comparison, etc. We would like to compare * to products like OnDo (www.brekeke.com), vegastream's (www.vegastream.com) Vega 50 FXS/FXO and Nortel's Option 11c mini. OnDo's offering is software based, vegastream's is an appliance and Nortel's is traditional. What is *'s competitive advantage to each of these types of offerings? Cost and/or RAD with the LAMP stack? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and DTA-310 was [ MGCP w/8x8 DTA-310 and as5300 pstn gateway]
- Original Message - From: Eric Wieling To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 11:07 AM Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway The Packet8 8x8 DTA-310 that I have ran SIP when I was using it. Eric, did you get the DTA fully working with Asterisk using SIP? Do you know what firmware you were using and can you post your sip.conf. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Changes (NAT-SIP)
I am having the same problem with a new CVS. Patrick also has the problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035114.html Keven had a problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035262.html but was able to get it fixed. Can you post a patch?. My asterisk computer is multi-homed behind NAT so maybe that is a factor? Is Asterisk behind NAT working with a new CVS for anybody? Thanks, - Original Message - From: Asterisk User Group [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 10:16 PM Subject: [Asterisk-Users] CVS Changes (NAT-SIP) I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can audio streams go client to cleint with IAX?
With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can audio streams go client to cleint with IAX?
- Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 02, 2004 5:27 PM Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. Thanks for the reply. Can you tell more about that last statement. If the audio doesn't serparate from the call control can the server keep track of how long the clients stay connected? Can it see DTMF that is sent between the clients and act upon it? Does IAX do this by default of do you have to set a parameter in IAX.conf. Thanks again ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wav49 voicemail problem with Windows Media Player
I have done some more investigating and posted this in Bug Tracker I have found that the Microsoft Sound Recorder will play the original posted wave file msg.WAV without errors. I opened this file and then re-saved it inside of Sound Recorder with the same GSM 6.10 (wav49) format. The resulting file (msga.WAV) is slightly different than the original. The msga.WAV file plays without error on Windows Media Player. Maybe this will give someone a hint as to what the problem is. - Original Message - From: Sean Cheesman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 11:15 AM Subject: RE: [Asterisk-Users] wav49 voicemail problem with Windows Media Player I am having problems too Just shy of the 5-second mark in the test vm. WMP 9.00.00.3075 Windows 2000 SP4 -Original Message- From: Warwick Ward-Cox [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wav49 voicemail problem with Windows Media Player I'm having the same problem. Warwick - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 5:39 PM Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wav49 voicemail problem with Windows Media Player
Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 and lawsuits
Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 12:12 PM Subject: Re: [Asterisk-Users] 911 and lawsuits I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Agreed, Guess I should have said traditional computer. Most PBXs would only use a hard drive for voice mail. A hard drive failure would not cause the PBX to stop working. Also, with something like Asterisk that is changing so often, there is always the possibility of a typo that is not discovered until you need to use one of those rarely used features like calling 911. Most business would have lots of cell phones around but in many metal building they do not work. They also don't provide the address information that a land line phone provides. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits and redundancy
- Original Message - From: Jonathan Moore [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 12:34 PM Subject: Re: [Asterisk-Users] 911 and lawsuits and redundancy This is esp true of any VoIP PBX system. In fact I think many of them run Windows. Or VOIP in general. This is what Vonage makes you agree to in their Terms of Service. 2.4 Requires Activation: You acknowledge and understand that 911 dialing does not function unless you have successfully activated the 911dialing feature by following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. You acknowledge and understand that you cannot dial 911 from this line unless and until you have received a confirming email. 2.5 Failure to Designate the Correct Physical Address When Activating 911 Dialing: Failure to provide the current and correct physical address and location of your Vonage equipment will result in any 911 communication you may make being routed to the incorrect local emergency service provider. 2.6 Requires Re-Activation if You Change Your Number: You acknowledge and understand that 911 dialing does not function if you change your phone number unless and until you have successfully activated the 911 dialing feature following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. 911 dialing must be re-activated. Although you may have activated 911 dialing with your former Vonage phone number, you must separately activate 911 dialing for any new number. 2.7 Change of Physical Location of Vonage Equipment: You acknowledge and understand that 911dialing does not function properly or may not function at all if you take your equipment with you away from the address or physical location that you have designated. 2.8 Requires Re-Activation if You Move: You acknowledge and understand that 911 dialing does not function properly or at all if you move or change the physical location of your Vonage equipment to a different street address, unless and until you have successfully activated the 911 dialing feature following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. 911dialing must be re-activated although you may have activated 911 dialing using your former address, and you must separately activate 911 dialing for any new physical address. Failure to provide the current and correct physical address and location of your Vonage equipment will result in any 911 dialing you may make being routed to the incorrect local emergency service provider 2.9 Possibility of Network Congestion and/or Reduced Speed for Routing 911: Due to the manner in which it is technically possible to provide the 911 dialing feature for Vonage DigitalVoice at this time, you acknowledge and understand that there is a greater possibility of network congestion and/or reduced speed in the routing of a 911 communication made utilizing your Vonage equipment as compared to traditional 911 dialing over traditional public telephone networks. You acknowledge and understand that 911 dialing from your Vonage equipment will be routed to the general telephone number for the local emergency service provider, and will not be routed to the 911 dispatcher(s) who are specifically designated to receive incoming 911 calls at such local provider's facilities when such calls are routed using traditional 911 dialing. You acknowledge and understand that there may be a greater possibility that the general telephone number for the local emergency service provider will produce a busy signal or will take longer to answer, as compared to those 911 calls routed to the 911 dispatcher(s) who are specifically designated to receive incoming 911 calls using traditional 911 dialing. 2.10 Automated Number Identification: At this time in the technical development of Vonage 911 dialing, it may or may not be possible for the Public Safety Answering Point (PSAP) and the local emergency personnel to identify your phone number when you dial 911. Vonage's system is configured in most instances to send the automated number identification information; however, the phone system routes the traffic to the PSAP and the PSAP itself must be able to receive the information and pass it along properly, and they are not yet always technically capable of doing so. You acknowledge and understand that PSAP and emergency personnel may or may not be able to identify your phone number in order to call you back if the call is unable to be completed, is dropped or disconnected, or if you are unable to speak to tell them your phone number and/or if the Service is not operational for any reason, including without limitation those listed elsewhere
Re: [Asterisk-Users] no monthly fee
http://www.iconnecthere.com and http://connect.voicepulse.com as long as you don't need an incoming phone number. - Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 10:34 AM Subject: [Asterisk-Users] no monthly fee Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can X100P detect phone pick up like an answering machine
If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to the pass-through port of the X100P? I know some PC software with voice modems can do this, just wondering if X100P/Asterisk can do it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
- Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, December 12, 2003 8:25 AM Subject: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) It would be great if the IAX protocol will be able to tranfer fax data (even converted in another format) between Asterisk boxes, using low bandwidth codecs like GSM. I know that this is possible only with the G.711 now (passing faxes using the audio stream), but maybe in the future...some native support will permit this. Sounds like you are thinking of something like T.38. T.38 is a FAX over IP protocol but my guess is that there would be licensing issues using it with Asterisk. It probably would not be that hard for someone to come up with a completely open FAX over IP protocol. A FAX over IP protocol would not have to operate in real time like VOIP. You could actually use acknowledge signals and retransmit packets that are dropped so that you get a perfect fax transmission. This can not be done with VOIP because by the time you realize you have missed a packet it is too late to re-transmit the packet. With FAX over IP it just means it takes a little longer to transmit the page. You could also design the protocol so that you could transmit at whatever rate you wanted. Slower links would use less bandwidth but take longer to send. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channelbank Recomendation and GS102 question
- Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 04, 2003 7:54 PM Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. Can I ask what Telephone/Internet service provider you are getting this from? Does anybody else have a setup like this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring new system for a non-profit organization
- Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 12:36 PM Subject: [Asterisk-Users] Configuring new system for a non-profit organization Hi, The PBX at the Colorado Organization for Victims' Assistance fried as a result of the building power being cycled. I'm now in the process of building an * system to replace the failed PBX. Minimum cost is the priority. I have a T100P card installed in the new system, and I am about to order integrated T1 services from the CBeyond company. They will require eight voice channels and at least eight data (they presently have DSL.) The rep. says that for the same cost ($520/mo) they will get all 24 channels as data--the channels are dynamically allocated. As each voice call is initiated, a channel will be pulled from data and used for voice. Can the T100P handle this dynamic allocation? Or must the channels be fixed? Thank for your help, Michael Welter Michael, unless Cbeyond is doing something different now, they provide a Cisco IAD as part of their service. The IAD has an Ethernet port for your Internet connection and POTs ports to connect to your PBX. My guess is that you will have to put a channel bank between their IAD and your T100P. By the way we had the Cbeyond service and it worked very well. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceGlo
- Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 2:14 PM Subject: RE: [Asterisk-Users] VoiceGlo Hi, Anyone knows what USB phone are they using? Where can one get it from? http://www.voiceglo.com/pages/Products_equipment.html Thanks! Its a Cyberphone K http://www.voipvoice.com/products/cyberphonek.asp Available here for $62.65 http://shop.voipvoice.com/buy/products/CyberPhoneK.asp but I think you can get it cheaper from voiceglo even if you just sign up for their $12.95 plan for one month and then cancel. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceGlo
- Original Message - From: Gary [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 9:07 PM Subject: Re: [Asterisk-Users] VoiceGlo which would make their Multimedia Terminal Adapter an interesting device ?? Interesting yes, but it does not support IAX. It is made by Innomedia http://www.innomedia.com/products/mta3000/mta3328_features.html My guess is they are using SIP with the MTA to talk to Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring power on Analog adapters
Anybody have any ideas why this fax machine won't work with any analog adapter I've tried? Have you verified you have the right tip-ring polarity? Maybe this is one of those few devices that it makes a difference when it is backwards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandsteam to support iLBC
Since quite a few people in the Grandstream improvements. thread have requested support for other low bandwidth codecs. I thought I would post this link. http://www.globalipsound.com/newsroom/releases.php?newsID=46 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk behind NAT?
- Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 14, 2003 7:28 AM Subject: RE: [Asterisk-Users] Running Asterisk behind NAT? Is this not just a case of a new entry in sip.conf EXTERNIP = external IP with the code for the contact header modified to use it (if present). Then the external firewall is set to forward the rtp and 5060 to * .. I know many people either have sip aware firewalls (as i do) or their * box has a real IP, but the number of people requesting this feature seems to be growing by the hour. I'm trying to get this working for quite a number of FWD users, at the moment I'm trying to fudge it with partysip... it's not very pretty and requires a linux iptables based firewall it's not big, it's not clever and it's certainly not funny I think this is a good idea but at least for FWD users can't they just use the FWD proxy that is designed to handle clients behind NAT with no special software on the client. The ones that allows even Windows Messenger to work behind NAT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
Have you tried limiting your fax machines to a lower baud rate like 9600. I know on Vonage this seems to help. - Original Message - From: Eduardo Goncalves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 26, 2003 10:10 AM Subject: Re: [Asterisk-Users] Fax and SIP On Thu, 26 Jun 2003 09:01:21 +0200 Florian Overkamp [EMAIL PROTECTED] wrote: Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. I've tested with both on the localnet (same ethernet hub) and I still get errors on the fax machine. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8 VOIP service now 1/2 the price
Unlimited US and Canada VOIP to PSTN calls for $20/month no equipment fees, no contract Does anybody have direct Asterisk to Packet8 fully working without the MTA? http://biz.yahoo.com/prnews/030609/sfm088_1.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
See this for one review. http://lists.digium.com/pipermail/asterisk-users/2003-March/009446.html I think some other people have it working but I am not sure if they got all the bugs out. - Original Message - From: Richard Alexander [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 09, 2003 2:00 PM Subject: RE: [Asterisk-Users] Packet8 VOIP service now 1/2 the price Do Packet8 provide the necessary info to use it with * for inbound and/or outbound ? Any idea what codecs they support ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
I think this is the company that makes them but it is hard to tell. http://www.artech.com.tw/html/english/AX300/AX300.htm This company sells them http://www.aislecom.com/ A rep. for them posted this thread, claimed to be the manufacturer. http://lists.digium.com/pipermail/asterisk-users/2003-March/009134.html There are quite a few comments so click on Next Message There is someone on eBay selling them http://tinyurl.com/bp4x disclaimer: I have never used one. I am not associated with the seller. On other lists I did hear some people had problems with them. You may want to start another thread and ask if any * users are using them. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 2:10 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? Hi, I look for something in the price range of a X100P for one FXO port. regarding the Dlink device, I think that there is not a real FXO port, more somethink like in Actiontec's InternetPhoneWizard, just to be able to use the analog phones for both IP and PSTN calls. It just switch one of the phone t the PSTN line. Have you considered a S100U and one of those $35 FXS to FXO converters? There is something like that? Where I can find such a converter and how this thing works? BR, Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 6:34 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Review: Packet8's DTA310
Have you tried getting Asterisk to connect with Packet8 directly such as: Packet8 -- internet -- Asterisk -- DTA If so, did you get it to work? Thanks for the review. - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 28, 2003 12:42 PM Subject: [Asterisk-Users] Review: Packet8's DTA310 DRAFT DRAFT DRAFT DRAFT I've been using the DTA310 from Packet8.net for a couple of weeks. The DTA310 is about $130 without the Packet8.net VoIP service. It only supports SIP. On the back of the DTA310 is a power connector (power supply is provided with the product), a 10/100 Ethernet port, an FXS port, and a reset button. The front of the device has LEDs for Power, Link, Phone, and Message. The unit I received has a minor defect where the power connector is a little flaky, but I doubt that it's a general problem. There is no POTS line pass-thru port on the device. The unit supports static IP and DHCP configuration and has a decent web based interface for configuration. It also supports VLAN tags for QoS type of configurations. It supports simple dial plan configurations so you don't have to wait for a timeout or press # before the call is sent to Asterisk. I like this feature a lot. It supports Inband and RFC2833 signaling and allows you to squelch inband DTMF. It also allows you to suppress voice packets during RFC3288 events (I have no idea what this would be useful for). I have found no real documentation for the device. I was hoping for at least documentation for the wild-card characters in the dial-plan configuration. The unit supports G711 ulaw, G711 alaw, G723 (the default codec), G726, and G729. I have only use the G711 ulaw codecs with Asterisk. Others may work. It supports SNMP, a password for access to the web interface (no password by default) and time-zone configuration and NTP. The device supports Caller-ID. The only minor problems I've had with the device is that when I hang up from an inbound call sometimes Asterisk calls the device for some reason. It rings twice and then stops. Also the message waiting light doesn't work with Asterisk. I have a cordless analog phone plugged into the device and the message waiting light on the doesn't work either. If the device can't register with the SIP server you won't get a dial-tone and you can't seem to access the web based interface. Also if you do a factory reset (hold down the reset button while powering up the unit) all the codecs seems to be disabled. Using the web interface you can upgrade the firmware on the device via TFTP. I've not tried using the device with a NAT firewall between it and the Asterisk server. Packet8.net does NOT officially support the device except for connecting to their VoIP service. When I contacted them for help in getting the device to work with the Packet8 service when I moved it behind a firewall they were VERY, VERY helpful and responsive. They even asked for a copy of the config of my Cisco router (the device doing the NAT). The only time they didn't want to help is when I was trying to connect the device to my local Asterisk server. I can't really fault them for that. I've had no problems with the sound quality. Overall I'm quite happy with the device. --Eric Wieling ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Developers Kit with TDM10B ?
Mark, Is there any chance for a new Asterisk Developers Kit with one Wildcard X100P and one TDM10B at a bundled price? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
