Re: [asterisk-users] NuFone suggests to use Vonage!!!!
On Sun, 9 Jul 2006, Andrew D Kirch wrote: To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Some people are into SM I guess. We have used NuFone. No problems during that period. If you know what you doing it's not bad. regards joe ___ --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 IP500 Forward problem codec issue
:19:18 NOTICE[2185]: channel.c:1314 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729 since our native format has changed to ulaw -- SIP/---.---.241.35-4e1f answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered SIP/---.---.241.35-40400490 -- Attempting native bridge of SIP/---.---.241.35-40400490 and SIP/---.---.241.35-4e1f == Spawn exten (Charity, ---0059, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' */SNIP Im sure I could change everything to ulaw G711 the problem would go away but I do not want to do that. Any Ideas? Thanks Scott H -- Joe Baptista www.joebaptista.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN number with callhunt and voicemail we web interface
can anyone recommend a plain ol fashioned telco in north america where i can get a DID with call hunt and voicemail which can be programmed via web interface. thanks and pls pvt joe ___ 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] Problem with voicemailsystem
On Sat, 27 Nov 2004 [EMAIL PROTECTED] wrote: Does anyone knows a possibility to disable the message of the server and only able the message of our client? Example: client says:Im not in my office, please leave a message. Well, after this message the sever should send the signal and record the opposite, without the message... to leave your message please speak after... blablabla a fast and dirty solution is to replace the server message to leave your message with a the beep file. regards joe ___ 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] Is there alist of codec by asterisk version?
Is there alist of codecs asterisk actually has per version number - i.e. 0.7, 0.9 etc? On 9 Jul 2004, Wolfgang S. Rupprecht wrote: Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) very helpful wolfgang - thanks joe ___ 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] Slackware 10.0 and asterisk and 2.4 vs 2.6
I'm installing the new Slackware 10.0 distribution - but not sure if i should go with the 2.4 kernal - which i think is the default install - or the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal? thanks joe ___ 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] FYI House bill exports analog phone regs to VoIP
-- Forwarded message -- Date: Wed, 07 Jul 2004 00:31:21 -0400 From: Declan McCullagh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Politech] House bill exports analog phone regs to VoIP http://www.politechbot.com/docs/boucher.voip.bill.070604.pdf There's a new bill in the House of Representatives to regulate phone calls made over the Internet. It was introduced this evening by Reps. Rick Boucher, D-Va., and Cliff Stearns, R-Fla., and it's called the Advanced Internet Communications Services Act of 2004. I've placed the text online and have a summary and commentary at News.com. The AICS bill takes a more regulatory approach than a competing proposal from Rep. Chip Pickering, R-Miss. Boucher and Stearns say VoIP servcies shall be subject to access charges and universal service taxes -- which is a really big deal. More on this later. Bill text: http://www.politechbot.com/docs/boucher.voip.bill.070604.pdf News.com coverage: http://news.com.com/Congress+mulls+new+Net+phone+rules/2100-7352_3-5258191.html?tag=cd.top http://news.com.com/Can+VoIP+survive+Congress%3F/2010-1028_3-5256334.html?tag=st.rn ___ Politech mailing list Archived at http://www.politechbot.com/ Moderated by Declan McCullagh (http://www.mccullagh.org/) ___ 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] VoIP under attack ... Bellcos rock and roll out PR
There has been a sudden increase in VoIP pro and con articles. Lots of political discussion. I think we are seeing the first major PR attempts by Telcos to stem the VoIP tide. regards joe ___ 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] Randy Bush ?-) intel - telco contract? time will tell.
On Tue, 6 Jul 2004, Kevin Walsh wrote: People are entitled to ask questions; If no questions were asked then this mail list would not have the volume of articles that it has. Absolutly correct - except for Randy who has a tendancy of starting arguments over irrelevant trivia. My own concern is that this list does not degrade into some DNS government mailing list full of trolling. I've seen randy post here many times. Not a problem - but his recent posts smell of trolling. Certainly arrogance and attitude were in abudant evidence. But maybe i'm wrong and just a bit too sensitive based on past experiences. We'll see. One of the things I appreciate in this forum is that people communicate and support each other - and they do a good job of it. We all share the same agenda - bury the telcos and move on. I know randy well - hes intel - would not surprise me if he was here with an agenda on behalf of the spooks he works for and their telco investments. After all he's never participate in anything unless he's gettin paid - randys no fool. but i could be wrong - time will tell. and as far as i'm concerned i really dont' think we should stretch the subject further cheers joe ___ 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] Re: iax or sip
On Mon, 5 Jul 2004, Randy Bush wrote: i did not criticize the protocol. remember, my question started with i am looking at iax to see if it is applicable to my needs. i don't need nats, nat traversal, nat anything. if i did, iax might well be one of the technologies i would consider. but i don't. Your up to no good again. Trying to troll? Look Bush it pretty easy - if you want a PBX use asterisk if not go away. regards joe ___ 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
IS VONAGE LISTENING? RE: [Asterisk-Users] Vonage and Asterisk integration
On Tue, 29 Jun 2004, Jay Milk wrote: Like I said, they just seem to be lazy and/or badly organized. If they can do LNP, why can't they change a hardline into a softphone, break one number out onto a different ATA, etc? I basically laid it out for them, saying If you can't move my 2nd line from this ATA to a new ATA, then I'll need to cancel that line... I no longer have that line. Not being able to something this simple cost them over $500/year from me... I wonder how many other Vonage users will drop them because of such things. We were considering Vonage - but if this is the case - we have no interest. Is vinage listening? regards joe -Original Message- From: Steve Kalcevich [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vonage and Asterisk integration Jay Milk wrote: I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. I never worked with vonage, is there tech support that bad? -- Regards, Steve Kalcevich, ___ 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] I need DIDs in Canada and USA with roll over option
I need a provider of DIDs with multiple inbounds. regards joe baptista ___ 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] Skype 4 Linux
On Wed, 23 Jun 2004, Stefan de Konink wrote: Hi All, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation layer)). It is a bit of work, but reverse enginering is too :) Just write some code we can stick in asterisk. ___ 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 emergency service and VoIP
I understand that most VoIP providers allow for 911 calling but that 911 service is not the same as that available to PSTN. From what I understand a 911 Call Will Go To A General Access Line at the Public Safety Answering Point (PSAP). This is different from the 911 Emergency Response Center where traditional 911 calls go. Does anyone know how I can get information on howto contact the people at the Public Safety Answering Points (PSAPs)? Is there alist somewhere I can reference. thanks joe baptista ___ 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] * as conference server for shoutcast.
On Sat, 12 Jun 2004, Jeremy McNamara wrote: Joe Baptista wrote: i.e. iax1---+ iax2---| iax3---|-- * -- MeetMe -- shoutcast radio ---| iaxn---+ This already exists today in Asterisk: show application ICES excellent - but i get this when i run the command on console pmg*CLI show application ICES Your application(s) is (are) not registered i'm using * 0.9.0 - i assume it's available in a later version. or can i impliment with 0.9.0 if i follow bartons guidelines at http://www.voip-info.org/wiki-Asterisk+cmd+Ices p.s. everytime i find something new about asterisk - i fall in luv. god bless you guys for makin it possible. regards joe baptista ___ 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] * as conference server for shoutcast.
i.e. iax1---+ iax2---| iax3---|-- * -- MeetMe -- shoutcast radio ---| iaxn---+ would be a great way for corporation to broadcast meetings. board meetings live management meetings live sales demonstrations and all interactive. you get the picture. right now i'm thinking of just patching a cable from the * operator console to a shoutcast server. thank god for old tech. but i would luv to explore the potential to feed an mp3 stream to a shoutcast server. * is mp3 complient. i'd be willing to use my test system - which is currently called frankenstein and for good reason - to experiment. just an idea to the * community. regards joe baptista ___ 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] any banks or financial institutions using asterisk
I've been approached to research and develop a system using asterisk. It will be used mainly to provide voice support to about 10,000 IAX clients operating on bank ATMs. So was wondering if there were any financial institutions, banks etc. using * and any comments would be much appreciated. regards joe baptista www.joebaptista.com www.baptista.god ___ 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] dialplan experts needed
I have the same situation - i.e. three different extensions scattered about. But I don't try them each individually. When a call comes in my asterisk attempts to ring up to four different devices at the same time. To do this using your dial plan is easy - i.e. exten = 555,1,Dial(SIP/1000SIP/2000SIP/3000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup All the phones will ring at the same time and the phone to pick up first wins. regards joe On Mon, 7 Jun 2004, Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ 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] FireFly - no sound after first call
I've been watching to see if this problem comes up with anyone elses firefly - but so far i'm the only one experiencing the problem. When I connect to either my asterisk server or FWD all goes well on the first call. I can hear and talk. But every call after the first one I end up with no sound - not even ringing. I use win98 and have tried it on two systems with win98 installed. thanks joe p.s. if any other info can provide let me know. ___ 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] e164.org
On Sun, 23 May 2004, Tony Hoyle wrote: Simon Dorfman wrote: I wonder if someone can help me understand this. Let's say I configure my asterisk box to use e164 and then I try to call a phone number in Germany. I'm in the U.S.A. So if the number I'm calling in Germany is registered in e164's dns, would my call be routed directly via their voip provider? Or directly to their asterisk box? And would it be free? From the looks of it, they're just a directory... it looks like their not running asterisk themselves. They use something called EnumLookup which I guess is some kind of plugin/script. If the number you're calling is in their database, it calls the VOIP number directly, otherwise it calls the POTS number No they just provide the dns - it's your equipment that does the connecting. Exampple - if you use my number 17058721310 and conver it to enum format it will look like this: 0.1.3.1.2.7.8.5.0.7.1.e164.org If you do a look up of that number - you'll see the associated enum pointer. example: dig 0.1.3.1.2.7.8.5.0.7.1.e164.org. any ; DiG 9.2.2 0.1.3.1.2.7.8.5.0.7.1.e164.org. any ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: NOERROR, id: 44741 ;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 9, ADDITIONAL: 4 ;; QUESTION SECTION: ;0.1.3.1.2.7.8.5.0.7.1.e164.org.IN ANY ;; ANSWER SECTION: 0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN TXT Joe Baptista 0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN NAPTR 100 10 u E2U+IAX2 !^\\+17058721310$!iax2:[EMAIL PROTECTED]/17058721310! . ;; AUTHORITY SECTION: e164.org. 600 IN NS alberta-2.bcwireless.net. e164.org. 600 IN NS ns1.au1.com.au. e164.org. 600 IN NS ns1.e164.org. e164.org. 600 IN NS ns2.au1.com.au. e164.org. 600 IN NS ns2.au1.net. e164.org. 600 IN NS ns3.bcwireless.net. e164.org. 600 IN NS apollo.bcwireless.net. e164.org. 600 IN NS mutual.bcwireless.net. e164.org. 600 IN NS alberta.bcwireless.net. ;; ADDITIONAL SECTION: ns2.au1.net.172800 IN A 202.87.28.2 mutual.bcwireless.net. 3600IN A 198.231.65.11 alberta.bcwireless.net. 3600IN A 209.115.243.234 alberta-2.bcwireless.net. 3600 IN A 66.244.202.59 ;; Query time: 343 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sun May 23 10:15:37 2004 ;; MSG SIZE rcvd: 434 In this case systems doing enum lookups are told to contact th iax2 server at phone.joebaptista.com use a guest login. So basically enum is a facility which uses the dns to route calls in such a way that they bypass the regular PSTN system if an internet routing and supported protocol exists. for more information on ENUM visit http://www.rfc-editor.org/rfcsearch.html and search for enum and e.164. Now the proper domain for enum is e164.arpa - and the respective assignments by the ITU of telephone delegations under e164.arpa has been at best very lame. As you can well imagine telephone companies have as a rule refrained from adopting e164 because as I have mentioned above e164 facilitates the bypassing of PSTN via the internet and that would of course mean lost revenue. Check out http://www.itu.int/osg/spu/enum/ for more info on the politics and bureacracy of enum. You can get information on enum assignments and registries at: http://www.ripe.net/enum/request-archives/ At you can see there are very few official enum registries - so I think what e164.org is doing in invaluable to the community. regards joe baptista ___ 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 ENUM service, what do you think?
On Sat, 1 May 2004, Dean Collins wrote: Yes but no information about how this will operate, what regulation or restrictions on joining, what connection protocols will be used etc etc agreed - you see alot of business fluff - but the technicals are very important and on many of these ventures they fail to include them. regards joe www.baptista.god Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reid A. Forrest Sent: Saturday, 1 May 2004 8:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New ENUM service, what do you think? From http://www.thevpf.com/ To join, please e-mail [EMAIL PROTECTED] or telephone 1-212-232-2020 (Mon-Fri 9AM-5PM EST). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jimfl Sent: Saturday, May 01, 2004 5:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New ENUM service, what do you think? Jim/frank, Can you give us more information about how to access this enum? I've been to the stealth web site and there is no information about access. I look forward with interest to what you have up and running today for asterisk users to benefit from. Cheers, Dean Sorry, I am not associated with Stealth in any way. Just saw the news story and thought it would be of interest to Asterisk users. It sounds like you don't have to be a VOIP provider to get access to their service. They talk about businesses using the service. If anyone finds out how to get access to their service, please post. Jim ___ 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
Re: [Asterisk-Users] Using IAXTel to dial FWD
On Sat, 1 May 2004, Rich Adamson wrote: now. But if you have a look at this page - http://www.freeworlddialup.com/advanced/iax you will find that you can now use FWD with IAX2 along with SIP :) FWIW, I just moved our FWD account to iax2, and it works rather well with *. The referenced web page does have a couple of configuration errors on it, but nothing all that difficult to diagnose/fix. Also, it appears the FWD - IaxTel definitions are incorrect (again) causing problems with connectivity in both directions. Could you share your conf files with us. I can connect to FWD on my asterisk - but FWD only see me as an external SIP agent and not a SIP client of the FWD network. DOn't know exactly why - so would luv to compare your conf files. thanks joe Joe Baptista: USG Portal www.joebaptista.com, Personal www.baptista.god LOW: low cost, Low Lands. http://www.dot.low/ Everything you need to know about LOW ___ 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
