Re: [asterisk-users] les.net losing DID's
That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
If the provider is selling the service and you are paying for the service the provider should give you the best service. If the provider can't give you the BEST service at that price then the provider SHOULD charge more and not waste my time. The providers are charging LOW PRICES to get customers that they can't handle at that price and they are not going to stay around because they can't pay to keep there server on line so when the provider goes under then the customers that paid lets say $50 - $200 like some did with Sunrocket then everyone will loss. The bottom line is low prices are good but to low is bad. What is a TO LOW per min rate? You tell me. Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Anthony Francis wrote: You know the problem is that most consumers think that it is possible to get the best and the most reliable for almost nothing. They go out with this expectation and get the cheapest, then when it bites them a few times, they scream why me. -- Original Message -- From: SIP [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sun, 05 Aug 2007 19:49:40 -0400 Worthless comes in many forms, Doug. If you're talking specifically about the monetisation of hardware/effort, then it may indeed be worthless by the simple fact that the cost may outweigh the net gains in profits gained from the purchasing, configuration, and deployment. Businesses are about making money first and foremost. If the amount of time and money put into a particular project outweighs the money you get in return, it's a bad business decision. Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000763-5, 08/05/2007 - 8/5/2007 8:14:47 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Providers
Anthony, So you know all 4 that work at teliax.com I only know what others have told me about teliax.com Most of what I know was told to me from someone that worked there. Best regards, Al Bochter http://www.BochterServices.com --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Anthony Francis wrote: Darrick Hartman (lists) wrote: [EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. I also vouch for Teliax as I send overflow LD through their trunks. I know the people there and they are great guys. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000757-4, 07/18/2007 - 7/19/2007 10:35:00 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Info about Providers
To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fairly open and truthful And this list will be carefully moderated Please do some posting! By the way I am looking for moderators for the list if you want to help let me know. -- Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Negotiation
So who do you pay to use the G723 codec? Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- O.Kamal wrote: I am having a problem with my asterisk gateway, it is accepting only G729, the client is offering G729 and G723.1, however for some reasons, around 15% of calls are rejected due to failed codec negotiation giving an codec error No compatible codecs, not accepting this offer. Anyone gone through this before? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000756-0, 07/12/2007 - 7/12/2007 2:21:04 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Visually impaired employees
I have a customer asking about the type of equipment there is for visually impaired employees working in a call center for inbound sales. -- Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID Spoofing to be banned in the USA
Well the gun owner will go to jail! Take a look at your local news. Best regards, Al Bochter Bochter Services --- Need to call use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Business Opportunity Click Below (9 Min Call) http://www.bochterservices.com/voip/iaxphone.php?cn=18003946919 --- Tim Panton wrote: On 28 Jun 2007, at 17:42, J. Oquendo wrote: Dean Collins wrote: Anyone running caller id spoofing applications in the USA running asterisk? Then it’s time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing- about-to-be-outlawed.html Why it means nothing... You're a carrier doing VoIP... Say a managed carrier. You re-sell trunks. One of those trunks maintains their own PBX. PBX admin decides to spoof out and is using a proxy say in India. Hell make it Tor for that matter. What's to prosecute? Prove it happened from where you say it did - remember the burden is on the prosecution. Now as the carrier (me) first thing I'm going to do is track down which trunk it came from... Then go to that client... So what happens if say the client was legitimately owned and had various proxied addresses committing toll fraud. Analogy... Gun dealer sells a .45 to an authorized gun buyer. Gun owner leaves his gun at home. Someone breaks into his home, cracks his gun safe, uses his gun for a crime, re-enters and places the gun back in the safe. Now its known it wasn't the gun owner because he was witnessed by the court system and recorded say at jury duty... What do you do, prosecute him? For what? Negligence? It would be humorous to see how this plays out. To me its more or less voting time let's sign pretend laws for brownie points The situation here in the UK is that the folks who interconnect to the PSTN have to validate that you own/control the number you are sending via IAX or SIP. We had a problem where an internal id was not getting overwritten with a valid PSTN number, one of our suppliers set a default caller-id and another rejected the calls. The process is annoying, but it works fine, you have to either use callerids of DIDs you have bought from the same ITSP or fax them a telco bill indicating your rights to that number. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000752-7, 07/01/2007 - 7/1/2007 12:26:58 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID Spoofing to be banned in the USA
I think you should be able to spoof your caller id to a number you are in control of. Like a toll free number, your main inbound and/or a number that goes to that ext. I think it is a big pain that anyone can spoof your cellular number and if you don't use a password can check your voicemail. How I read the upcoming law that is how it is going to be that you can spoof to a number that you are in control of. And I am fine with that. On the Asterisk server we use I have one inbound trunk that our toll free rings to and 4 outbound trunks that have no caller to them there are not any DID set to them. So for my outbound what would my provider set my caller ID to? Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Stephen Bosch wrote: Al Bochter wrote: Well the gun owner will go to jail! Take a look at your local news. If you own a gun, it's your responsibility to keep it secure. I don't know of an OECD juridiction where that's not the case. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000752-7, 07/01/2007 - 7/1/2007 2:16:13 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zlib1g
Tzafrir Cohen My advice: If the information is outdated Submit updated information Best regards, Al Bochter Bochter Services -- Need to call use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Tzafrir Cohen wrote: On Wed, Jun 20, 2007 at 03:32:19PM -0700, bilal ghayyad wrote: Dear Cohen; In this link: http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html In the subject: 2.Installation, then in the sub title: Zaptel Installation Please advise. My advice: don't use obsolete doucmentation. That incorrect recommenndation is not the only mistake in that page. ___ --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] Que on A2Billing
What is the point of line lights on the phone? The lights are so you would know when the KSU is out of lines. With Asterisk if the system is setup right it should never run out of lines to use. Best regards, Al Bochter Bochter Services -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- John Novack wrote: Given that Asterisk is modeled on, in the telephone industry, an obsolete PBX design, without many of the modern day hybrid features, and only recently has any effort been made to provide buttons and lights for lines ( Is that yet working in 1.4??) one would have to do some very careful number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: Thanks man, Is there any other way without dialing 9... it will be kinda pain for a customer to dial 9 every time and plus they need to know also... Is there any intelligent way to identify? if its a local SIP then don't route to Trunk else route to Trunk. Cheers, Nitesh Guillermo Salas M. wrote: On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote: Thanks man... So far everything worked as expected... How can I make internal calls stay within the PBX. For example, when one SIP-Friend tries to call another SIP-Friend without sending the call out on Trunk and receive it back. Same like dialing from one extension number to another extension. My SIP-Friends are using US DID numbers and I would like to keep the local calls within the network. Right now when I try to call other SIP-Friend, I get a message saying The number you have dialer is currently not available... while the SIP-Friend is registered. Try dialing the number 9 before the sip/iax2 friend number. Regards, Cheers, Nitesh ___ --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 Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 1:47:44 PM ___ --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] CNAM.
If you want to look up phone numbers try and its FREE http://www.asteriskextras.com/index.php?option=com_contenttask=viewid=21Itemid=2 Best regards, Al Bochter Bochter Services -- Did you check your US Greenbacks for GOLD Today? http://www.bochterservices.com/?t=USbill_email -- James FitzGibbon wrote: On 6/17/07, *Nick Seraphin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Yes... 1.5 cents per dip... you prepay the fees... and they deduct from the prepaid amount. You can start with $5.00 which seems like a low-risk to check it out at least. The CLEC I use is more expensive that that for CNAM, and they want to do it on EVERY incoming call, even wrong numbers, whether it's answered or not, per PRI. So since I get several thousand wrong numbers a month, and only 100 or so calls that I actually CARE what the CNAM is on those calls, I can set it up in Asterisk to only do the dip for certain DNIS numbers. I calculated that instead of $70+/month this will cost me $1.50/month. Nice savings. :-) I just hope it's reliable when the call volume picks up more. I gave this a shot yesterday. I figure I can stand to lose $5 if it sucks. Which for someone in Canada, it does. Granted, their website is somewhat hazy on whether or not they support Canadian CNAM - part of the page says can I look up numbers outside the US and Canada while part says outside the US, then the body says we don't support non-NANPA numbers. Pretty much every number I have tried to look up so far for Toronto/GTA just gives me back the city for the name, so I get a bunch of NORTH YORK ON and TORONTO ON or CELLPHONE ON results back, but no actual names. I've gotten a few correct hits back on company numbers, but just as many wrong ones. The Hilton in Edmonton's number comes back as GTCO CALCOMP, and a company I deal with in Mississauga (in the 905 NPA) comes back as ETOBICOKE ON (which is in 416). On the upside, it did find PIZZA PIZZA correctly. /sigh Of course, this is all via their web portal. I am completely unable to connect via their AGI port as provided in their sample configuration page. I get connection refused, which under a stock 1.2 Asterisk drops the call, so I can't leave the dialplan logic intact in the hopes that this is a transient error. Attempts to telnet to the port given via their portal are met with an immediate RST packet, suggesting that their fastagi service is down. At least the cost to play was cheap. IMO, it's not ready for production usage (at least under 1.2 - under 1.4 you can recover from a failure to connect to an AGI service and continue dialplan execution) -- j. ___ --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 Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 1:31:40 PM ___ --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] Que on A2Billing
In a2billing just change the 9 to what you need it is right in the conf file. Best regards, Al Bochter Bochter Services -- Need to call me use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Nitesh Divecha wrote: Thanks everyone for the input... In real world we can not ask the customers to dial 9, if they want to call another SIP user... and trust me its confusing for a customer also... meaning when to dial 9 and when to not... We have a custom proprietary system which does this part very well... Before it sends the call on a Trunk it will check the DID, if it exists within the local system. If it does then it will just use IP to IP call, else send the call to Trunk... I think its possible to do this by creating some basic dial plans... Same like creating local extensions. Cheers, Nitesh John Novack wrote: Given that Asterisk is modeled on, in the telephone industry, an obsolete PBX design, without many of the modern day hybrid features, and only recently has any effort been made to provide buttons and lights for lines ( Is that yet working in 1.4??) one would have to do some very careful number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: Thanks man, Is there any other way without dialing 9... it will be kinda pain for a customer to dial 9 every time and plus they need to know also... Is there any intelligent way to identify? if its a local SIP then don't route to Trunk else route to Trunk. Cheers, Nitesh Guillermo Salas M. wrote: On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote: Thanks man... So far everything worked as expected... How can I make internal calls stay within the PBX. For example, when one SIP-Friend tries to call another SIP-Friend without sending the call out on Trunk and receive it back. Same like dialing from one extension number to another extension. My SIP-Friends are using US DID numbers and I would like to keep the local calls within the network. Right now when I try to call other SIP-Friend, I get a message saying The number you have dialer is currently not available... while the SIP-Friend is registered. Try dialing the number 9 before the sip/iax2 friend number. Regards, Cheers, Nitesh ___ --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 Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM ___ --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] Semi OT - Cable products suppliers
Graybar is high priced you can do better anywhere. I only use Graybar when no one else has what I local and I need the parts A.S.A.P Best regards, Al Bochter Bochter Services Did you check your US Greenbacks for GOLD Today? http://www.bochterservices.com/?t=USbill_email Nick Seraphin wrote: My favorite place for all cable infrastructure products is the local Graybar warehouse. I'm lucky to have one only about 20-25 minutes away. Check www.graybar.com to see if they have one near you. On the web, www.ablecomm.com has some nifty and hard to find telecom-related products and tools, but they are expensive. I've ordered from them several times... they're reputable. But expensive. Graybar is the cheapest place I've found. If someone knows a good web-based store with better pricing and good selection, I'd love to see it for my own use. :-) -- Nick On Fri, 8 Jun 2007, [EMAIL PROTECTED] wrote: Anyone have a good recommendation for a supplier of punch blocks, 25 pair connectors and cables, etc? Thanks BEN BROWN ___ --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 Inbound (clean). Database: 000747-1, 06/05/2007 - 6/8/2007 12:49:12 AM ___ --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] click to call
Nick You made a very good point. Best regards, Al Bochter Bochter Services Did you check your US Greenbacks for GOLD Today? http://www.bochterservices.com/?t=USbill_email Nick Seraphin wrote: On Sat, 2 Jun 2007, Steve Totaro wrote: That is a totally different concept than we have been discussing. You are talking about actual phones and the person clicking, then entering their phone number having to pick up a physical phone. This is as trivial as generating a .call file and dialplan magic. The concept we are discussing is clicking a link that connects the clicker to whatever via the computer using a headset or speakers and a mic. No phone or numbers involved, at least to the clicker. The problem is, the only people who will be able to use that link are geeks that have a headset/mic on their computer. Most normal people don't have those devices, and even if they did, they feel much more comfortable with the concept of making phone calls using a telephone. We all often forget that the vast majority of the outside world is not technically-inclined in any way, and that unless your web site is only targetted towards computer geeks, you're creating a huge barrier for the average customer. Everyone has a phone, though. If the analog FXS adapter had not been created and reduced to an affordable price, VOIP would still only be about as popular today as it was in 1995. -- Nick ___ --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 Inbound (clean). Database: 000746-2, 06/01/2007 - 6/2/2007 8:02:20 PM ___ --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] Trixbox problems
Dave Please note what the core is.. * Asterisk(tm) Open Source PBX The GUI only writes some of the conf file for you. So if there is a fix for the list member that works on Asterisk please help them out. I have worked on other Asterisk based PBX systems and the conf files are just about the same. I am not saying Trixbox is better just easier for the new guy Me I don't like GUI's I prefer the hard way. That way I know what conf files do what to the system and that makes it easer to fix latter. * HUDLite server/admin (via package manager) Just slows the systems down and I see no good use for HUDLite Yes Trixbox does have alot of USELESS Packages added on to it. But keep in mind it is still Asterisk based at the core. The bottom line is. - Trixbox is still [asterisk-users] Best regards, Al Bochter Bochter Services Did you check your US Greenbacks for GOLD Today? http://www.bochterservices.com/?t=USbill_email Dave Cotton wrote: On Tue, 2007-05-15 at 19:16 +0200, Dave Cotton wrote: Perhaps the fancy-shmancy GUI is hiding the configs. Al Bochter has just told me off list that Trixbox is Asterisk But according to their site trixbox is a complete application platform. When you install trixbox you have a powerful application platform at your fingertips. Products included with trixbox include: * trixbox dashboard * Asterisk(tm) Open Source PBX * FreePBX web management tool * SugarCRM * Munin (via package manager) * HUDLite server/admin (via package manager) * IVRGraph (via package manager) * phpMyAdmin? (via package manager) * Webmin (via package manager) So I still wonder if the GUI hides the configs. As I've mentioned on Talkshoe my GUI is called vi. ___ --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] While the VoIP-Info.org site is down...
So does anyone know when Voip-info.org will be back up? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Shane Breen Sent: Wednesday, March 14, 2007 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] While the VoIP-Info.org site is down... Feel free to use: http://www.thetelecomdirectory.com/forum If you register your company here as well: http://www.thetelecomdirectory.com You will be able to upload white papers, list your company in our directory, release press releases all for FREE. Here is where you do all of the above: http://www.thetelecomdirectory.com/signup/signup.asp If you want to see how The Telecom Directory ranks visit: http://www.alexa.com/search?q=thetelecomdirectory.com Hopefully VoIP-Info will come back up but in the meantime use the site to its full potential. IT IS FREE. - Original Message - From: Matt Riddell (NZ) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 14, 2007 4:22 PM Subject: Re: [asterisk-users] RE: what happened to asterisk wiki??? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 JR Richardson wrote: A friend of mine was on the site yesterday, late morning, when he refreshed his screen, a banner came across the web page VOIP SUCKS and then the site was no longer available. I'm pretty sure the site was compromised by some hacker trying to prove a point or make a statement. Not to throw stink on anyone or group, but maybe it was someone from a competing open source VoIP project or one of the Big Iron VoIP System Manufacturers. Probably just some cracker with too much time on their hands. I feel like someone shot my dog, please get the site back up as soon as possible. There was a post about a security vulnerability in wiki on bugtraq a couple of days ago, but it looked more like someone had figured out how to edit pages (pointless considering a wiki is open anyway). - -- Cheers, Matt Riddell Director ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67 NV4lAmOkaISR79fBTjajGw8= =u7sc -END PGP SIGNATURE- ___ --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 Inbound (clean). Database: 000723-2, 03/14/2007 - 3/14/2007 6:33:59 PM ___ --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: Sending SMS
Steve Well I have 300 for $5.00 thats .016 cents each IF I use all 300 now if I go over I pay .15 each ( I Think) Never went over I see that the cell providers are looking at SMS as internet data over there system and I do agree that there is more money in data than voice services. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Gordon Henderson wrote: On Fri, 2 Mar 2007, Al Bochter wrote: I don't see why the cost to send SMS is around .15 each. What does the gateway know that I don't know about sending the SMS. I just think .15 for each SMS send is high. Or am I just over looking something? You're missing nothing; The telcos have us by the short curlys. For them, it's money for old rope. They probably (in the UK at least) make many times more money through TXT messages than voice. The base rate here is about 12p a message. 12p for 160 bytes, or a single data packet over their network - which would be over £700 per MB. There are now bolt ons or additional packages depending on the network you're with - eg. with my contract I get up to 500 free TXTs a month. I know some people who send dozens a day here. (Especially young people - I think most 10 year olds now have mobile phones!). It's scandalous, but no-one challenged it when they first anounced it because we all thought it was fantastic! The best thing they ever did was for the 4 networks (in the UK) to agree to pass TXT messages between each other. That was some 6 or 7 years ago, maybe more, and that's when it really took off big time in the UK. I doubt it'll ever change because that's the way it's always been, and no-one is going to challenge them in a serious fashion. (And no-one else can afford to build up a network to make it possible!) I've not really looked into the TXT sending business via landline in the UK, but I think it's basically a call to an 09xxx number - which are premium rate numbers, charging up to £1.50 a minute. Lets hope the 160 byte packet gets sent in less than a minute! The stats. are amazing too. I looked at wholesale connection last year for a project. They had rates of up to a million messages a month. (do the sums and workout how many miuntes there are in a month...) A quick search shows that in 2004, we in the UK were seding over 20 billion TXT messages a year - Thats 75 million a day. Not bad for a population of 65 million... Who knows what the rate is today... http://www.theregister.co.uk/2004/01/22/uk_text_message_volumes_break/ Ah, that was 2004. Looks like we're almost doing that per month: http://www.theregister.co.uk/2006/06/26/uk_sms_record/ 3.3 billion texts sent in May 2006... Gordon Text messaging is not that big in the US for some reason. Well anyways, on my T-Mobile phone, I have an unlimited text message package that cost $15/mo. I am not sure how many constitutes unlimited though, I have not read the small print. Thanks, Steve ___ --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 Inbound (clean). Database: 000721-1, 03/03/2007 - 3/3/2007 9:05:20 AM ___ --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: Sending SMS
I don't see why the cost to send SMS is around .15 each. What does the gateway know that I don't know about sending the SMS. I just think .15 for each SMS send is high. Or am I just over looking something? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Tomislav Parcina wrote: Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 I don't see a point of using providers as Bayhamsystems. First, it's unpractical to send SMS from phone. If I'm going to use web interface, then is better to use some provider that has web interface just for that (or maybe they will provide application to send messages to groups or in certain time). Only reason why I would like to do it true Asterisk is if I could use my VoIP or E1 provider so that I get only one bill. But using Bayhamsystems that isn't a case. So, why people use such providers? ___ --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] Sending Email From the dialplan
I have looked around with no luck. Does anyone know of a way to send an email from the dialplan. The system that I am working on has none thing to do with VoiceMail. This is something like the SMS command but using sending email I am working on a prepaid alarm dispatch program for Asterisk if anyone has any input please let me know. I will be more than happy to write the code as Open Source for others to use code. With help from the list. -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email ___ --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] dial a pager and enter DTMF
Buy a cap code from the paging provider and program that cap into the group of pagers that way when you page that cap code all of the pagers will trip. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Supa wrote: is there a way to pipe the dial command with SendDTMF(123456) What I am trying to do is dial an extension and have it page a group of pagers with the same number. Saving a lot of time over dial each one manually by hand. ___ --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 Inbound (clean). Database: 000716-3, 02/23/2007 - 2/24/2007 9:12:09 PM ___ --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] Sending SMS
Is there anyone sending SMS with Asterisk? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email ___ --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] chan_sip.c:1968 create_addr: No such host:
Its not right. I am using a2billing calling card and it works fine Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email If you need to contract Customer Service Please use our IAX2 WebPhone at the link below http://www.bochterservices.com/voip/iaxphone.php [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Inbound (clean). Database: 000714-3, 02/18/2007 - 2/18/2007 10:42:53 AM ___ --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] chan_sip.c:1968 create_addr: No such host:
nope Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email If you need to contract Customer Service Please use our IAX2 WebPhone at the link below http://www.bochterservices.com/voip/iaxphone.php Rob Hillis wrote: I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Inbound (clean). Database: 000714-3, 02/18/2007 - 2/18/2007 10:43:30 AM ___ --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: Fwd: [asterisk-users] Some queries on g729 license.
Derek Whitten Messages like this SHOULD NOT be posted to the list I have been trying to block you from my servers do to your abuse I will add this email address to the list also and contract your service provider. You are not doing the right thing you are acting like a child. I think you are abusing the list to send SPAM. And it is getting old blocking your email addresses And it getting old that you spoof my mail server and sending email with that look like it is coming from my servers. Derek if you keep this up I will press charges on you. I do track IP address on all email to my servers so yes I have all the proof I need from you. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Derek Whitten wrote: C F wrote: I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Jan 8, 2007 8:22 PM Subject: Re: [asterisk-users] Some queries on g729 license. To: [EMAIL PROTECTED] (C)UNT (F)UCK! THIS IS OFF THE LIST FUCK YOU ASSHOLE! GET A JOB AND STOP LIVING OFF MY TAXES YOU DON'T KNOW WHAT YOU ARE DOING TRY AND STAY ON THE POINT. YOU ARE NOW BLOCKED I AM NOT GOING TO DEAL WITH JACKASSES LIKE YOU GOOD BYE Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: When I first noticed that this thread has over 20 messages i was sure it is interesting. When I read it I realized that I havn't noticed that Al Bochter has posted to it. Plain old stuff, just someone making sure to put a new twist on it. On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote: The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have
Re: Fwd: [asterisk-users] Some queries on g729 license.
David So do you think Digum and Sipro is now one in the same code with G729 in mind? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email David Thomas wrote: This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Regards, David ___ --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 Inbound (clean). Database: 0702-0, 01/09/2007 - 1/9/2007 5:23:48 PM ___ --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] Some queries on g729 license.
What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] Some queries on g729 license.
Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: What about the free open source G729 To use a g729 codec you must pay a license fee to the patent holder. It is immaterial as to whether the implementation is open/closed source. ___ --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 Inbound (clean). Database: 0701-4, 01/08/2007 - 1/8/2007 2:46:30 PM ___ --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] G729 license counting
You need a license when ever you transcode the audio From any codec to G729. or G729 to any codec you will need a license for each instance. If you call into your system from a provider that uses G729 you don't need a license If you check your voicemail that is saved on your system in GSM format then you need a license to transcode the file from GSM to G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Douglas Garstang wrote: That's not correct. You need one G729 license for each transcoding instance. If you have two SIP channels and both are G729, then no license is required. If you have two SIP channels, and one is G729 and the other is ulaw, then a license is required. Doug. -Original Message- From: Zoa [mailto:[EMAIL PROTECTED] Sent: Monday, January 08, 2007 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 license counting Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 2:47:33 PM ___ --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] Some queries on g729 license.
Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? As long as you don't transcode than 10 at a time. Is that legal? I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 Legal Stuff - Important, please read To use G.729 or G.723.1 _*you may need to pay a royalty fee.*_ Please see http://www.sipro.com for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. To distribute Intel's IPP libraries with a commercial product, you may need to pay a once-off license fee to Intel (currently $US180). My patches to Intel's code are distributed free under the GPL. Most of the code is just Intel's sample code re-arranged a little bit to work the way Asterisk expects. Therefore, this work would not have been possible without Intel doing 99.9% of the work. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Please read the entire page. From the link you sent: Why NOT G.729? There are some reasons you might /not/ want or need to use G.729. * You don't want to pay the license fees or use the codec without the permission of the patent holder. ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 3:24:08 PM ___ --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] Some queries on g729 license.
Mike What I was looking to do is use the easier to install one and the better one. I was asked by a customer about using G729 and I told the customer that they would have to pay for the G729 licenses. The customer pointed out the open source G729 code and I was not sure if I could use that. Then I was told by others that work on Asterisk that the open G729 was a cracked ver of Digum G729 and don't use it without buying the Digum licenses. So that is what I am tring to found out. And Paul did point that out that the open G729 and Digums code is not the same. I don't have Open G729 or Digum G729 installed in the Asterisk server. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Mike wrote: Al Bochter wrote: Mike I understand that. but it states on there site and note the key words may need What I want to know is if you buy 10 licenses from digum can use the Open Souce code? That is not what you said or asked. You were asserting that a free as in beer solution existed. If something says may it is incumbent upon you to decide if the rules/requirements in question are applicable to you, nobody else knows your situation. To answer your new question, as I am not an expert in patent law I haven't a clue. I see the note about the IPP license From what I have been told this is easier to get working than Digum's G729 I use Digium's codec and found it very easy to install. ___ --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 Inbound (clean). Database: 0701-6, 01/08/2007 - 1/8/2007 3:57:10 PM ___ --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] Some queries on g729 license.
Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] The Good, Bad and Scam VoIP Providers
So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --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] The Good, Bad and Scam VoIP Providers
If anyone would like to read more or the links Contract me off the list And my point about that message was NOT about Bret it was all bad providers in general Do should do your homework on who you are giving you credit card info to. That was my point of The Good, Bad and Scam VoIP Providers I am sorry that I named a provider. But let that go and get to the point of my massage The Good, Bad and Scam VoIP Providers So yes the point, the boat, and other form.. was totally missed Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Tom Lynn wrote: And it seems likely to me that you'll be sued for libel. On 12/24/06, *Al Bochter* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com http://trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 11:32:44 AM ___ --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] The Good, Bad and Scam VoIP Providers
So I will try get you on my point of the message! So if you get a VoIP Provider that states on there web site that they will give you unlimited use for $7.95 per month. You start to use there service and your VoIP service stops working after 20 days and you contract the provider and the provider states that you ran over you limit for the month.. You tell the provider that you had the unlimited plan for $7.95 per month Then the provider states well unlimited is only 1500 minutes per month Now is a my point of The Good, Bad and Scam VoIP Providers And I never named any providers and I do have a few. Definitions of *unlimited* on the Web: # having no limits in range or scope; to start with a theory of unlimited freedom is to end up with unlimited despotism- Philip Rahv; the limitless reaches of outer space # outright: without reservation or exception # inexhaustible: that cannot be entirely consumed or used up; an inexhaustible supply of coal Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email ___ --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] The Good, Bad and Scam VoIP Providers
Peter Bowyer I understand that. But from my standing that would be a scam and a rip off And I do understand that the providers are buying minutes in bulk When the provider states unlimited then the service should be without bounds Keep in mind there other things that the VoIP Providers are doing is low down Please lets stay on my point.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: On 24/12/06, Al Bochter [EMAIL PROTECTED] wrote: So I will try get you on my point of the message! It would appear to be 'unlimited doesn't mean unlimited'. Surely this doesn't come as a surprise to someone who has been in the industry as long as you claim to have been? Move on, nothing to see here. Peter ___ --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] The Good, Bad and Scam VoIP Providers
Peter // I'm done with this. I thought we were discussing VoIP provider scams? You are the one posting massages that are off the subject I took your replys off the list. Please keep your posts on the subject ( Thank You ) :-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: Oh no, the game's up - Al's found my IP address. Wait - no he hasn't - he's found an IP address that belongs to McAfee Security in Spain - with whom I have no connection at all. (Hint: whois ip address) Those PI classes really paid off, Al. Supposing you had managed to find out one of my IP addresses (which isn't really too hard, I have NIC handles at ARIN and RIPE, and hold addresses on behalf of more than one major organisation), what were you going to do with it? I'm done with this. I thought we were discussing VoIP provider scams? On 24/12/06, Al Bochter [EMAIL PROTECTED] wrote: Peter, This is off the list? it looks like ip: 62.189.112.129 Country GB: Britain AM I close? Anyways This is off my point! And should not be posted to the list. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: This is getting funnier by the minute. Way to go, Al. On 24/12/06, C F [EMAIL PROTECTED] wrote: I Find It Funny, So I Decided To Let Others Laugh As Well -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Sun, 24 Dec 2006 14:01:06 -0500 Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: [EMAIL PROTECTED] This is off the list C F, You are an ass Bret is a scammer you can take that to the bank from a PI. Sorry I never stated what I do for a living. Did I? I will be dealing with Bret. And 2007 is not going to be a good year for that scammer. So why are you hiding use a real email address. And a real name. Looks like you have an in with Bret Master of Cybercrimes May have to my homework on you to. What is you think? I really don't care if you if you trust me. Your reply is only a pop out trying to save your ass. Please stay on the POINT! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: Al, Nobody Cares About Your Problems With Bret. Most People Here Know And Trust Bret More Than They Do You. All You Have Done So Far Is Made A Fool Out Of Yourself. At This Point All I Can Think Of Is That If Bret Does Hold Some Of Your Money That It Is A Significant Amount And He Wont Ever Give It To You. Move On And Dont Make A Bigger Fool Out Of Yourself. Swallow Your Pride Its Not Fattening. For You I Can Say: Temper Is What Gets You Into Trouble Pride Is What Keeps You There. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --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 Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 1:41:46 PM ___ --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] The Good, Bad and Scam VoIP Providers
Steve Can you please stop posting messages to the list that don't have anything to do with VoIP or the subject. I took the replys to your messages Off The List (Thank You.) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Pulled from Junk Folder Yes, I filter by your name in the body as I know it will be Junk. Thanks, Steve SYSOP wrote: Did you filtered this one to junk? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Sorry folks, not much traffic on the list today and I want to expose this guy for what he is. Al, you are already filtered to junk, so no need for the autoresponse. Thanks, Steve Al Bochter wrote: Steve This is off the list This is off the point also :-) I am going to setup an auto reply do to dumb asses like you. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: What a tool. Al Bochter wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 3:46:49 PM ___ --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 Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 3:57:48 PM ___ --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] The Good, Bad and Scam VoIP Providers
Brian Capouch I changed the subject I don't think it was right for this message!! // Re: [asterisk-users] Need quality toll free 800 number over IAX? Well I don't agree with you about this thread they are talking about the good and the bad VoIP providers This is information that Asterisk users MUST KNOW. We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) The BAD VoIP providers must try to get there servers and customer service right or they need to go way. Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Brian Capouch wrote: Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. ___ --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] The Good, Bad and Scam VoIP Providers
Tzafrir Cohen, Well if you would have asked I don't aim to sell service to VoIP users. I BUY VOIP TRUNK SERVICE from VoIP Providers. I BUY VOIP DEVICES from suppliers I install Asterisk PBX Servers and point the my customers to VoIP Providers and Suppliers So the fact is I don't offer a competing service. I sell the services to the END USER. Like a supplier said to me once. I will take care of a contractor before I return a call to an End User. The End user is only one sale and alot of time. The happy contractor's are 100's of sales The other way to look at this is the contractor / installer is 100's of end users So what I stated has everything to do with that. If I point a client to a BAD VoIP provider or supplier that make me look bad. And I could lose sales So Trafrir what do you do? I looked at your site it look like you would be a VoIP SUPPLIER? So you are a supplier competing for sales from myself and others on the list? If you would make a note I changed the Subject line I started a new Topic. So I am on-topic. Bad service is a big deal so the tone should be VERY LOUD. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Tzafrir Cohen wrote: On Sat, Dec 23, 2006 at 05:30:54PM -0500, Al Bochter wrote: Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services And the fact that you offer a competing service naturally has nothing to do with that. So please keep your tone down and stay on-topic. ___ --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] The Good, Bad and Scam VoIP Providers
Steve Totaro, I will contract you off the list about trxtel that is not my base point of this. // Bottom line, you get what you pay for. I agree. // Check out a provider, try their customer service, see if there is a toll free number, call it and see if someone picks up. You forgot word of others that used the server. // Use whois to see how long they have been around, ask questions, and use common sense. It is called due diligence. Whois.org is not going to tell you much about them. Ask questions? HM MM is that not what I am stating here?? And have others tell you how the provider was to them. Sorry you trying to shoot me down on that point. / / This is information that Asterisk users MUST KNOW. /is simply not true. // Expand your horizons, expand your vision. Do not automatically assume that everyone using Asterisk is using a VoIP provider. So are you stating that if the provider (ANY POT or OTHERS) gave you bad service you would stay with them and not tell anyone. // Post to the biz list where this belongs. What am I trying to sell??? This is end user stuff Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Steve Totaro wrote: I seriously doubt trxtel.com scams anyone. I may be wrong but the person behind it has been with this community for a long time and has done nothing post insightful and meaningful things to this list and give back to the community in many other ways as well. It is a unique idea but that is really all I know about it (the service). I fire customers all the time. I would probably fire you if you were my customer based on the way you are ranting. In these cases, the drain is not worth it personally or for the business so bye bye. Bottom line, you get what you pay for. Check out a provider, try their customer service, see if there is a toll free number, call it and see if someone picks up. Try it over and over. Use whois to see how long they have been around, ask questions, and use common sense. It is called due diligence. As for me, I use Asterisk in a very LARGE (although everything is relative) deployment but I use no VoIP providers. I terminate to a T3 (28 T1s), all PSTN ULAW. The only VoIP that we do is INSIDE ONE DATA RACK and is traditional telephony one form or another outside of that rack. /This is information that Asterisk users MUST KNOW. /is simply not true. Expand your horizons, expand your vision. Do not automatically assume that everyone using Asterisk is using a VoIP provider. Post to the biz list where this belongs. Thanks, Steve Al Bochter wrote: Brian Capouch I changed the subject I don't think it was right for this message!! // Re: [asterisk-users] Need quality toll free 800 number over IAX? Well I don't agree with you about this thread they are talking about the good and the bad VoIP providers This is information that Asterisk users MUST KNOW. We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) The BAD VoIP providers must try to get there servers and customer service right or they need to go way. Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US. SO the providers and suppliers need to get there acts together. The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW PBX CUSTOMERS TO VONAGE. Is this what the list wants I DON'T THINK SO Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Brian Capouch wrote: Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. ___ --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 Inbound (clean). Database: 0662-0, 12/22/2006 - 12/23/2006 6:27:07 PM ___ --Bandwidth and Colocation
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
You guys are missing the point of the message I sent! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Bill Hackensack wrote: Geez Al, let it go. We've heard your rants for what seems like years now (even though it's only been weeks). No one cares anymore. ___ --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 Inbound (clean). Database: 0662-0, 12/22/2006 - 12/23/2006 9:13:10 PM ___ --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] IAX connection to FWD not working
The same with our servers. I just deleted the FWD trunk. That took less time and quit using the FWD Account If anyone has any info on why please let me know. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Timothy Parez wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 Inbound (clean). Database: 0660-0, 12/19/2006 - 12/20/2006 3:24:32 AM ___ --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 quality toll free 800 number over IAX?
I have used www.ipkall.com I have had one way audio for two weeks now with no reply from CS. So I will back you up on this I guess http://www.kall8.com/ would be the same I think they are one in the same. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Kevin Walsh wrote: www.IPKall.com [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. ___ --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] DTMF Tones A-B-C-D
Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email ___ --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] DTMF Tones A-B-C-D
Are you sure IdsFisk will do all 16 DTMF tones? I have the free ver of IdeFisk and that one does only the base 12 DTMF tones Base 12 DTMF are 1 2 3 4 5 6 7 8 9 0 # * The 16 DTMF are 1 2 3 4 5 6 7 8 9 0 # * A B C D If the paid ver of IdeFisk has that may have to pay the money but first I must know for sure. :-) I want to use A B C D for control IVR's Not Everyone knows about the 16 tones like the Hams do 8-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Zoa wrote: Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it). Zoa Al Bochter wrote: Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D ___ --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 Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM ___ --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] DTMF Tones A-B-C-D
Thanks Bob I will have to download the updated ver. then Don't mind me I had a brain fart.. :-[ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Bob Chiodini wrote: The free version 1.31 has all 16 keys on the keypad. Bob... Al Bochter wrote: Are you sure IdsFisk will do all 16 DTMF tones? I have the free ver of IdeFisk and that one does only the base 12 DTMF tones Base 12 DTMF are 1 2 3 4 5 6 7 8 9 0 # * The 16 DTMF are 1 2 3 4 5 6 7 8 9 0 # * A B C D If the paid ver of IdeFisk has that may have to pay the money but first I must know for sure. :-) I want to use A B C D for control IVR's Not Everyone knows about the 16 tones like the Hams do 8-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Zoa wrote: Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it). Zoa Al Bochter wrote: Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D ___ --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 Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM ___ --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 Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 4:05:10 PM ___ --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] Repeated Digits
I am experience repeated digits when connecting a call from SIP using any codex I have tried the same things to fix this. If anyone knows why please let me know. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Gustavo Flores wrote: Hi, Have anyone experience repeated digits when connecting a call from SIP and terminating it to a PRI Channel? On the other side of the PRI Channel is an IVR that expect a pin but the digits come repeated. For example, you dial 12345 but it is received as 12224445 ___ --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] Good Commercial Grade Service Provider?
I tried to setup an account with Cyberdyne-ip.com after filling out the form all I get when I try to log in is Invalid User name and password please go back javascript:window.history.back(); and try again If the login don't what about there service? :-\ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email William Piper wrote: Check out www.cyberdyne-ip.com http://www.cyberdyne-ip.com. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, *Paul Connolly* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com http://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 Inbound (clean). Database: 0659-1, 12/16/2006 - 12/17/2006 11:54:14 PM ___ --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] Good Commercial Grade Service Provider?
Ok I retyped the same information in same user name them tried to log in and it worked that time. But anyways am in.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email William Piper wrote: Al, I just logged in with _your_ username password and it worked fine for me. I used Internet Explorer and Firefox... both worked fine. I'm guessing that you may have typed in your password wrong. Please contact [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] from the email that you signed up from and we will forward your login info to you. Thanks, bp On 12/17/06, *Al Bochter* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I tried to setup an account with Cyberdyne-ip.com http://cyberdyne-ip.com/ after filling out the form all I get when I try to log in is Invalid User name and password please go back and try again If the login don't what about there service? :-\ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email http://www.bochterservices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email http://www.bochterservices.com/?j=goldt=email William Piper wrote: Check out www.cyberdyne-ip.com http://www.cyberdyne-ip.com/. Great rates, great quality, unlimited channels, and an easy to use GUI to manage your account. FYI, You may have more responses if you ask the -biz list. bp On 12/15/06, *Paul Connolly* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0659-1, 12/16/2006 - 12/17/2006 11:54:14 PM ___ --Bandwidth and Colocation provided by Easynews.com http://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 Inbound (clean). Database: 0659-1, 12/16/2006 - 12/18/2006 12:11:15 AM ___ --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] enum
use dundi Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Khaled wrote: Dear Please how can I make a local dns naptr on my system ,ro resolve local calls using enum Regards * 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 Inbound (clean). Database: 0658-1, 12/14/2006 - 12/15/2006 4:11:15 AM ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P gsm, ilbc, g729 etc are a lot better choice. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Steve Edwards wrote: This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The busiest hour will account for 10% of the daily total. This is how I'm figuring it... Casual observation shows that SIP setup and teardown takes about 26 UDP packets. Assuming the packets are full (512 bytes) this adds up to about 13 kilo-bytes for each call. I've heard that ulaw (including overhead) is supposed to take about 80 kilo-bits/sec. Assuming each call takes 10 minutes, each call will take 13 kilo-bytes + (80 kilo-bits * 60 * 10) or 48.13 mega-bits. Assuming (to make the math easier) 10 bits = 1 byte, each call will take 4.813 mega-bytes. So, 100,000 calls of 10 minutes (1 million minutes) would consume 481,300 mega-bytes per month or 3,333 calls consuming 16,043 mega-bytes per day. Assuming the busiest hour accounts for about 10% of the daily total, that hour would consist of 333 calls consuming 1,604 mega-bytes. So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Would anybody venture an estimate of what the peak bandwidth would be if we changed to IAX? With trunking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Inbound (clean). Database: 0659-0, 12/15/2006 - 12/15/2006 9:47:48 PM ___ --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] Input on Dundi
Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security ___ --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] Input on Dundi
Douglas. I can't agree more. Thats VoIP things for you little to no documentation :-( Well thats ok, I am working on some documentation for Asterisk and other Distros. a2billing is one I am working on dundi will be next. And others I will post the links when its ready and right. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Douglas Garstang wrote: It's just a shame there isn't complete documentation available. -Original Message- *From:* Bruce Reeves [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 12, 2006 9:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Input on Dundi I use it to handle calls between multiple sites connected over a wan. It works great, I finally understood the concepts after the Astricon presentation on clustering with dundi. On 12/12/06, *Al Bochter* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock http://www.bochterservices.com/?t=TFdid For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email http://www.bochterservices.com/?j=PBXt=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security http://www.bochterservices.com/?j=storet=email_security ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex 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 Inbound (clean). Database: 0657-0, 12/12/2006 - 12/12/2006 11:34:57 AM ___ --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] Vonage SIP access via asterisk?
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security BerkHolz, Steven wrote: Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA www.hirotecamerica.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 Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM ___ --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 freezes when DNS not working: a BUG??
Just do a lookup for the domain name and resolve it to the IP address Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Giorgio Incantalupo wrote: Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP so I cannot use it. Is there anybody who had a problem like this? 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 Inbound (clean). Database: 0654-0, 12/06/2006 - 12/6/2006 4:49:00 AM ___ --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] any possibility of Vonage Integration
I found the link for Vonage Integration with Asterisk http://www.vonage-business-plus.com/ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Vijay Gandhi wrote: Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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 Inbound (clean). Database: 0653-2, 12/04/2006 - 12/5/2006 12:58:38 AM ___ --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] any possibility of Vonage Integration
The providers have in there minds that A residential will use less line time than business will use. Like it was said I guess they don't have teenage kids There is more usage on my residential line than there is on my business line. I Put 1800 Mins on the cellular and about 1000 on the VOIP ( TOTAL = 2800 ) that is BUSINESS The house had an easy 4500+ mins this is RESIDENTIAL And I don't use the house line I can't ever get a Dial Tone just kids talking. Or the provider changed the dial tone sound to kids and wife talking :-\ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Lacy Moore - Aspendora wrote: On 12/6/06, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Time Bandit wrote: The TV ads promote it as unlimited. If there are real cases where residential subscribers did not get unlimited residential service for the advertised price, why aren't any state attorney generals going after vonage? Vonage clearly states that unlimited is not unlimited (not in their commercials, of course). I didn't have a bit of problems finding it. Their unlimited for business seems quite a bit too low for me, but then again, that's just my opinion, maybe businesses no longer use phones. Some things are clear and some things not so clear. I couldn't find anything where specific limits on minutes in or out are stated. I think they try to limit the number of accounts cancelled strictly for high minutes. Accumulate enough of those and a smart class action law firm will be after you. Anyway, how can they determine a residential line is being used for business without invading subscriber privacy? ___ --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 Inbound (clean). Database: 0654-0, 12/06/2006 - 12/7/2006 1:31:41 AM ___ --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] any possibility of Vonage Integration
And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: 1) You can connect the vonage lines to an FXO interface. I have a customer who has the linksys router/ATA connected to FXO ports of his nortel meridian PBX switch. You might try that with digium cards, FXO port of SPA-3000 or some multiport FXO gateway. 2) Vonage softphone accounts work for incoming with asterisk. Absolute forwarding, busy forwarding and multiringing to the softphone is treated as free in-network calls. Vijay Gandhi wrote: To be more elaborate, i am using 10 vonage lines in my office, can i connect them all using asterisk, or is it possible to configure those accounts on asterisk instead of the linksys boxes i am using. Regards Vijay Gandhi -Original Message- From: Vijay Gandhi [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] any possibility of Vonage Integration Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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 Inbound (clean). Database: 0653-2, 12/04/2006 - 12/5/2006 8:25:26 AM ___ --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] any possibility of Vonage Integration
Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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] any possibility of Vonage Integration
But I never said ATA. I said you call Vonage and tell Vonage that you want to B.Y.O.D. there is a KEY you need Vonage to get you and install into Asterick for Vonage service to work. Buy like Brad said there are easier ways than Vonage. I am not downing Vonage I have and still use Vonage and never had an outage with them. Yes I did install Vonage into Asterick so I know what you have to do. Just getting the right information you need from Vonage is the hard part Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 10:23:36 PM ___ --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] any possibility of Vonage Integration
Please hold :-) Now you will listen to MOH for 4 days :-D By the way you forgot one thing.. The person you get can't speak English. :-( Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Henry.L.Coleman wrote: This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 10:23:43 PM ___ --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 there any Asterisk controllable thermostat?
I would really like to see some documentation also. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Matthew Rubenstein wrote: On Mon, 2006-12-04 at 00:58 -0700, [EMAIL PROTECTED] wrote: Date: Sun, 3 Dec 2006 23:04:52 -0500 From: Zeeshan Zakaria [EMAIL PROTECTED] Subject: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I am wondering if there is any such thermostat available which can be controlled from Asterisk. Trixbox comes bundled with xPl, which is a home automation network API that is also common to Windows XP. I haven't seen any documentation of how to actually use it (with Trixbox/Asterisk), but I would be very interested in seeing some, including examples and supported HW. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria ___ --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] any possibility of Vonage Integration
You can add Vonage accounts to your asterisk. The only account that Vonage will let you use is there Biz account higher rates. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Vijay Gandhi wrote: Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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 Inbound (clean). Database: 0653-2, 12/04/2006 - 12/5/2006 12:58:38 AM ___ --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] G729 Passthru?
I think you do need to buy the G729 for each call. If your system is using anything other than G729. That is the way I was told it works. But I don't use G729. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Matthew Rubenstein wrote: I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming for feeding back to the SIP carrier. Does that loopback work without a G729 codec on the server? If not, what would the codec actually do with the data it gets? A related issue is whether I can pre-encode recorded audio files with a G729 codec. So the server can send wakeup call messages to the SIP carrier without running the codec at call time, just sending the pre-encoded media to the SIP carrier. ___ --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: IAX access to FWD broken?
FWD works fine for me. I just set up a trunk in asterisk. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Jim Lawson wrote: Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim Timothy Parez wrote: I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. ___ --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 Inbound (clean). Database: 0651-2, 11/28/2006 - 11/29/2006 9:27:39 AM ___ --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 toll-free US did
What price range are you looking for. We have toll free's with NO MONTHLY FEES Please let me know. Contract 1 866 638 1254 EXT: 250 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's in stock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Vicky wrote: I am looking for a toll-free US 1800 DID which can be setup quickly . I have seen nufone there quality is very good but they charge for 30 seconds minimum ( others do 6/6 i guess ) . east coast gateway server preffered . . Plz lemme know if you have some suggestions i want it to be setup very quickly :) . Thx . ___ --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 Inbound (clean). Database: 0650-2, 11/23/2006 - 11/26/2006 6:20:35 PM ___ --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] Welcome to Join Asterisk MSN Groups!
Why would I want to join MSN groups then MS can't get an OS right! Now MS whats to do get into VOIP that will be a total messup. The thing is when MS will try to say that they asterisk. MS has no place anywhere around Asterisk. You will see what I mean just look at the bottom of MY website. I just wanted to put my .02 in about MS and VOIP Servers.. I know some will agree with me some will not. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email@ (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Mayson.Wang wrote: :), welcome to join MSN groups: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]! Add to your msn friend, and /help for help! Have a good time here ! ___ --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 Inbound (clean). Database: 0650-0, 11/22/2006 - 11/22/2006 7:52:22 AM ___ --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] Question on CDR Database
The CDR could be used by billing software not all billing soultions do there account that way. he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended I am not sure what you are asking. You can log just about anything you want you just have to Change or Make the program to do what you need. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Noc Phibee wrote: Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye ___ --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 Inbound (clean). Database: 0649-1, 11/18/2006 - 11/19/2006 5:31:28 AM ___ --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] AdvancedVoIP Billing ?
Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ --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 Inbound (clean). Database: 0649-0, 11/15/2006 - 11/18/2006 7:13:37 AM ___ --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] AdvancedVoIP Billing ?
Really? I am using a2billing to bill customers for Per min DID inbound to there IVR's, Voice Mail Box tracking, Billing users for outbound from Softphones And Calling Cards :() there is alot more but I don't want type the much... My god it's asterisk think outside of the box Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Noc Phibee wrote: Yes ;=) but a2billing it's for calling card ;) Al Bochter a écrit : Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ ___ --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 Inbound (clean). Database: 0649-0, 11/15/2006 - 11/18/2006 1:04:03 PM ___ --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] STUN with one public and one private IP?
I never said voxbox is better than trixbox. I said You like trixbox Should try voxbox. The link is: http://www.easyvoxbox.org/ Trixbox has good and bad points (loads from RPM's) Voxbox has good and bad points (Loads from source) I like source better than RPM's -- Thats me, but I do programming Both are ISO and run asterisk You tell me the better one. If you try both... Good thing that I only told you about two out of (Well god only knowns) I know ten diff installs off hand. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Zeeshan Zakaria wrote: You said voxbox is better, but even the link you gave for them didn't work. I googled, and apparantly links are broken on their website. ___ --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 Inbound (clean). Database: 0649-0, 11/15/2006 - 11/16/2006 11:30:14 AM ___ --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] POS Terminals
Why are you using VOIP for credit cards? You have the Internet look into a bank with a credit card gateway. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Christopher Aloi wrote: Hello List - I've got a question regarding POS terminal transactions (credit card machines, ATM, etc...). Currently we setup customers in the following manner: Customer Location -- Data T1 -- DataCenter - PSTN Termination We are currently using Mediatrix gear for fax transmissions from the customer location, but they don't seem to handle POS modem sales very well. Does anyone have any experience using POS terminals? Is something like an IAXy at the customer prem a good idea? -Thanks for any advice, -- -- Christopher T Aloi -- ___ --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 Inbound (clean). Database: 0649-0, 11/15/2006 - 11/16/2006 11:30:46 AM ___ --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: unable to get channel lock BAD BAD BAD
Trixbox has the same asterisk core... As asterisk... As EasyVoxBox... To back up my point I can download the backup file from one and install them on the others The ONLY thing is FreePBX I must play with the conf for that. So what is your point ?? Please do tell me what your point of 98% of the people here don't use Trixbox. the core is still the same. The other thing is I have not stayed up to date with this feed Re: unable to get channel lock BAD BAD BAD Please email me all the details you have off the list and I will see that I can do. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Eric ManxPower Wieling wrote: Deadlocks are not a config or Trixbox issue. Doug Lytle wrote: Tim Uckun wrote: Judging by the lack of response here it seems like this is broken and nobody knows how to fix it. 98% of the people here don't use Trixbox. ___ --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 Inbound (clean). Database: 0649-0, 11/15/2006 - 11/16/2006 3:48:40 PM ___ --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] SIP Ports (1000 to 2000 works)
Where is your DMZ pointed? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: Thereis definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: No 1000 to 2000 is not a typo. Well let me put some light on this.. If you goto http://www.ipkall.com/ and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID's As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from http://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.conf This is what I ran into today So I guess you are right... It's a free for all on ports. Makes things harder to do. I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted // by different vendors, though. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use:1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before"So is there any standard ports" Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter ___ --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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/14/2006 2:29:51 AM ___ --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] Problem with internet down
What are you using for your Internet connection? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:46:33 AM ___ --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] Problem with internet down
Well if you want to use VOIP you will have to get a better Internet connection. You can't do anything to the PBX Server to fix this. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: We have a link dedicated of radio. But that presents problems the times! Al Bochter escreveu: What are you using for your Internet connection? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:46:33 AM ___ --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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:55:33 AM ___ --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] SIP Ports (1000 to 2000 works)
I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] SIP Ports (1000 to 2000 works)
Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before "So is there any standard ports" Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM ___ --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] SIP Ports (1000 to 2000 works)
No 1000 to 2000 is not a typo. Well let me put some light on this.. If you goto http://www.ipkall.com/ and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID's As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from http://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.conf This is what I ran into today So I guess you are right... It's a free for all on ports. Makes things harder to do. I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted // by different vendors, though. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter ___ --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] STUN with one public and one private IP?
You like trixbox Should try voxbox. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Steve Sobol wrote: I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using STUN, and I understand I need a primary IP and an alternate IP to make STUN work. Well, I got STUN working here on amethyst.justthe.net, which has a bunch of available public IPs, but the Trixbox only has one public IP, and I have to request (and pay for) more IPs from the phone company if I need any more. And I'd really prefer that STUN be running in the office, and not on my personal server. So I'm wondering... I'm using stund from SourceForge. Is there any reason I couldn't give the Trixbox's public IP address as the primary and 127.0.0.1 as the secondary? I believe Asterisk is listening on the loopback interface... Thanks in advance, Steve ___ --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] DID billing with a2billing
Never mind I got DID billing to work with a2billing it was in the conf files needed retyped to the right info. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Al Bochter wrote: Can anyone tell me what I have to do to get DID billing to word with a2billing. I am thing it may be context ___ --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] DID billing with a2billing
Can anyone tell me what I have to do to get DID billing to word with a2billing. I am thing it may be context -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] VM Language
What is the best way to have the voicemail system and system do more than one language I know I have to have all wav, gsm files on the system. -- Best regards, Al Bochter Bochter Services (Voip PBX) Free World DialUp: 780217 EXT: 250 WebSite: http://www.freeworlddialup.com/ http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] SIP v IAX2
But how do you deal with the cable co blocking the ports you need for SIP? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally "arrived" and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so "cheap" calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --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 Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --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] SIP v IAX2
VOIP is NOT telephone so the FCC don't have anything to say about VOIP. Well not right now. But in CAN there are cable co. that block the SIP ports and there is an up charge for them to unblock SIP. Ask Vonage.. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Dean Collins wrote: FCC if you are in the USA. Simple. Otherwise find another broadband provider. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Al Bochter Sent: Thursday, 2 November 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 But how do you deal with the cable co blocking the ports you need for SIP? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Henry.L.Coleman wrote: Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally "arrived" and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so "cheap" calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --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 Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 8:23:19 AM ___ --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 Inbound (clean). Database: 0645-1, 11/02/2006 - 11/2/2006 9:44:20 AM ___ --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] Java Web Phone
Anyone know the cost? Best regards, Al Bochter Bochter Services (Voip PBX) Free World DialUp: 780217 EXT: 250 WebSite: http://www.freeworlddialup.com/ http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vladimir Montealegre Estailes wrote: Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar ms rpido en Internet.Tutopia es Internet para todos. ___ --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 Inbound (clean). Database: 0644-4, 10/31/2006 - 11/1/2006 6:24:48 PM ___ --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] No ring tone when using IAX
When I call from a softphone using IAX2 there is no ring tone This is the same if I call in to the IVR and press # and dial the stations ext number no ring tone And I get the same if I call in using a DID on an IAX2 trunk BUT if I use anything that is SIP I get the ring tone Softphone DISA Trunks Let me know what I should check. -- Best regards, Al Bochter Bochter Services (Voip PBX) Free World DialUp: 780217 EXT: 250 WebSite: http://www.freeworlddialup.com/ http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] DTMF detection problem in PABX trunk
Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --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 Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___ --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] SIP v IAX2
Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] Broadvoice incoming DTMF problems
dtmf = inband Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Kevin Kiely wrote: Is anyone having problems and Broadvoice with incoming DTMF not being recognized from a caller originating on the PSTN connection to Broadvoice? Broadvoice tech support confirmed this issue as a result of their carrier connections and suggested a work around in the dial plan(SIPDtmf). This does work but breaks DTMF for BroadVoice callers. ___ --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 Inbound (clean). Database: 0643-2, 10/24/2006 - 10/25/2006 1:05:14 AM ___ --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] Broadvoice incoming DTMF problems
That too. I never used Broadvoice but from what users have told me high priced poor service. There are better with no connect fees Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Dovid B wrote: Is anyone having problems and Broadvoice with incoming DTMF not being recognized from a caller originating on the PSTN connection to Broadvoice? This is the reason why I left them two months after I signed up with them. Broadvoice tech support confirmed this issue as a result of their carrier connections and suggested a work around in the dial plan(SIPDtmf). This does work but breaks DTMF for BroadVoice callers. Find a better carrier :) ___ --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 Inbound (clean). Database: 0643-3, 10/25/2006 - 10/25/2006 9:34:56 AM ___ --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] Where to best start looking for voicemail/moh sound quality problem?
/ I've tried setting QOS parameters for IPCop but I'm sure that had any effect. Keep in mind QOS in only good from your server to the cable modem. QOS don't count past the modem you CAN'T set QOS on the Internet Best regards, Al Bochter Bochter Services Toll Free: 866-638-1254 EXT: 250 Free World DialUp: 780217 EXT: 250 Cellular: 206-203-5801 http://www.BochterServices.com/?t=Email - - - - we BUY and sell Coins, Silver, Sterling Silver and Gold http://www.bochterservices.com/?j=goldt=email - - - - For new and used security items http://www.bochterservices.com/?j=storet=email_security - - - - 24kt GOLD PLATING http://www.bochterservices.com/?j=platingt=email - - - - Frank Tarczynski wrote: I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've tried setting QOS paramters for IPCop but I'm sure that had any effect. Frank ___ --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 Inbound (clean). Database: 0643-1, 10/23/2006 - 10/23/2006 4:49:47 PM ___ --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] G.729 operating on outgoing only
disallow=Ulaw in the trunk conf Best regards, Al Bochter Bochter Services Toll Free: 866-638-1254 EXT: 250 Free World DialUp: 780217 EXT: 250 Cellular: 206-203-5801 http://www.BochterServices.com/?t=Email - - - - we BUY and sell Coins, Silver, Sterling Silver and Gold http://www.bochterservices.com/?j=goldt=email - - - - For new and used security items http://www.bochterservices.com/?j=storet=email_security - - - - 24kt GOLD PLATING http://www.bochterservices.com/?j=platingt=email - - - - Joel Lansden wrote: Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the providers server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf file has both the remote server my calls come from, and the remote server we send calls to listed, with disallow=all then allow=g729, but only the outgoing seems to be doing what its supposed to. Any suggestions? Joel Lansden Solutions Architect [EMAIL PROTECTED] tel 205.533.2039 fax 866.602.9130 digitalparadisesystems http://www.digitalparadise.net Could it be any easier? ___ --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 Inbound (clean). Database: 0642-6, 10/22/2006 - 10/22/2006 10:17:02 AM ___ --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] Findme problem
Yes Please still post that one. I would like to see the code. Best regards, Al Bochter Bochter Services Toll Free: 866-638-1254 EXT: 250 Free World DialUp: 780217 EXT: 250 Cellular: 206-203-5801 http://www.BochterServices.com/?t=Email - - - - we BUY and sell Coins, Silver, Sterling Silver and Gold http://www.bochterservices.com/?j=goldt=email - - - - For new and used security items http://www.bochterservices.com/?j=storet=email_security - - - - 24kt GOLD PLATING http://www.bochterservices.com/?j=platingt=email - - - - Dovid B wrote: Ooops. I read the email wrong. The macro I created called one number. If the person didnt accept the call or if they didnt pick up then it tried the second person. Let me know if you still want it. Dovid ___ --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 Inbound (clean). Database: 0642-3, 10/19/2006 - 10/19/2006 10:44:05 PM ___ --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] Speed Dials
Can anyone tell me where the numbers are stored for the 3XX speed dials -- Best regards, Al Bochter Bochter Services Toll Free: 866-638-1254 EXT: 250 Free World DialUp: 780217 EXT: 250 Cellular: 206-203-5801 http://www.BochterServices.com/?t=Email - - - - we BUY and sell Coins, Silver, Sterling Silver and Gold http://www.bochterservices.com/?j=goldt=email - - - - For new and used security items http://www.bochterservices.com/?j=storet=email_security - - - - 24kt GOLD PLATING http://www.bochterservices.com/?j=platingt=email - - - - ___ --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
